similar to: Polycom does not register from outside to asterisk

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

I've got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format slin since our native fo rmat has changed to gsm I have set ALLOW=ALL in

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I am new to this list and have been using Asterisk as part of my research project for about 2 weeks now. I would like to get your thoughts on a scenario that I am attempting at the moment. I haven't had luck until now. In this scenario, I am trying to register a SIP user configured on the zoiper client installed on a laptop, which is on the same Local Area Network, with the

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice in both 2 and 3

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

Hello guys! We released a new ZoIPer BIZ BETA (version 1.03). You can experience better look and more advanced features. Finally MS Vista fans can also make use of it. Zoiper BIZ BETA is available free of charge from www.zoiper.com. There you can find out more about the improvements and features. We are also offering customization packages for ZoIPeR Free Windows. Zoiper is a multiprotocol:

Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:

On Tue, Dec 23, 2014 at 6:51 PM, Joseph <syscon780 at gmail.com> wrote: > > > Most cell phone don't have a USB port but you are correct, maybe I just need > IAX2 soft-phone like: > Zoiper - it works on most of the platforms. I think Zoiper registers > directly with Asterisk IAX2 (if configured) as an extension, isn't it? If your cellphone is capable of a Wi-Fi

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

Hello list, I would like to seek your expert opinion on a setup I am trying as part of my research. I have not been able to successfully make a call so far. In my setup, I use two laptops that are interconnected by means of a stand-alone IS1581 switch. Thus there is no LAN involved. I have assigned static IPs to the two laptops, say 10.0.0.1 and 10.0.0.2. I have installed Asterisk 1.6.2.6 and

I'm having a little problem with voicemail. Actually I'm not getting the ability to leave a voicemail-message. This is part of the dialplan : > exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1}) > exten => s,n,NoOp(${ARG1}@boxes) > exten => s,n,Voicemail(${ARG1}@boxes) > exten => s,n,Hangup() > exten => s,n,MacroExit This is the