search for: softclient

Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >&...

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005> context=default host=dynamic disallow=all allow=ulaw allow=alaw extensions.conf exten => 2005,1,Dial(SIP/${EXTEN},10,tr) exten => 2005,2,Congestion...

...stion which I'm sure has been asked before but my research has yet to find it. I have Asterisk running on a Linux server but in order to get it to connect I needed to punch a hole in my firewall on port 5060 for it to receive the registration responses from broadvoice. If I run sjphone as a softclient on my home PC I do not need to punch that same hole and it works just fine. I'm confused as to why. I compared the logs of the softphone compared to Asterisk and noticed some differences but simply cannot get Asterisk to work without the port forwarding. The phone doesn't have STUN setup...

...e .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson

Hi, my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when user and asterisk are on Local LAN. But my asterisk server is on public ip and users are behind nat, and so this method says unknow host when used on public asterisk server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel.

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick

Does anybody have a reasonable solution for an Outlook MAPI plugin that works with asterisk? At very least, I would like Asterisk to push incoming call information to the computer, which should then open an Outlook form, launch a web browser, etc. Beyond that, it would be cool to have Outlook initiate outgoing calls. Shouldn't be too difficult, and I know some of you are working along

In taking a cursory browse at Vonage's site today, I noticed they are now offering a soft phone. Has anyone had any experience using this? And does this possibly open new opportunities for using Vonage with Asterisk? Just thinking outloud on the list, soliciting thoughts and experiences from others. AJ

.... Guess how you could find such a software. You might search google with the following search term: sip-softphone file-transfer video The very first result for me is Messenger - SIP Softphone - Soft Client File transfer. IP Telephone, Do not disturb, busy & available status. softclient conferencing, Custom available/away status. video telephony ... www.eyeball.com/products/messenger.html - 22k - Cached - Similar pages and from the decription on the page there, this software does what you want. That was not too difficult, was it? If you wanted to find out if som...

...hy I don't get any SDP coming back. The rfc says the ACCEPT SHOULD be present so I'm thinking that is a Asterisk bug perhaps. In example 1 My own UAC code generated OPTIONS includes the Accept header yet still I see no SDP coming back from endpoints. I have tried using X-lite and PhonerLite softclients. I'm hoping there is a simple explanation or something I can do. Is Anyone able to query codec capability for any endpoints outside of a normal INVITE? I would like to know how you do so. Below is excerpt from the automatic OPTIONS query I see in the sip logs from setting verify=true. No Acc...

Dear all I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any soft phone which support this all feature ?? with asterisk Regards Satish Patel --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.

...ld find such a software. You might search google with the following search term: sip-softphone file-transfer video The very first result for me is Messenger - SIP Softphone - Soft Client File transfer. IP Telephone, Do not disturb, busy & available status. softclient conferencing, Custom available/away status. video telephony ... www.eyeball.com/products/messenger.html - 22k - Cached - Similar pages and from the decription on the page there, this software does what you want. That was not too difficult, was it? If you wa...

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

...Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 On Mon, Feb 15, 2016 at 2:53 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > >> Thanks for the mighty quick response, Joshua! >> >> I am using Zoiper on Linux softclient: >> REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 >> > > That's the request URI, not the Contact header. The Contact contains the > URI that the server should dial to reach the client. The full message would > be useful. > > Cheers, > > > -- > Joshu...

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip client to (an expensive) pstn destination. The CDR from my telco tells the same! First case: Sep. 05 2006 11:46:40 20 secs call from X to 0900xxx (valid call, micropayment) Sep....