similar to: Re: SJphone registration problem - Help!

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up

Not when I click on that link. On May 13, 2018 7:37:50 AM PDT, Rui Barradas <ruipbarradas at sapo.pt> wrote: >Hello, > >I don't understand. > >It *is* the same question. Same code, same words. And same 'AKSHAY M >KULKARNI' (the OP here) and 'AKshayKulkarni' (SO). > >Exactly the same. > >Rui Barradas > >On 5/13/2018 2:07 PM, Jeff

Hello, I don't understand. It *is* the same question. Same code, same words. And same 'AKSHAY M KULKARNI' (the OP here) and 'AKshayKulkarni' (SO). Exactly the same. Rui Barradas On 5/13/2018 2:07 PM, Jeff Newmiller wrote: > I am puzzled by the use of the term "cross-posted" here... I don't see the OP or their question or any similar words from the

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to use this account with AsteriskNOW to begin my experience with a VoIP based PBX. The

All, I am setting up my first * box and am trying to configure SIP Softphone to SIP softphone dialing. When I dial ext. 2001 it rings once (Very short) then immediately goes to voice mail for ext. 2001. I keep seeing the "Got SIP response 482 "Loop Detected" back from 192.168.1.252", immediately followed by "No one is available to answer at this time". I've

Ah... Looks like I have the 500's...Sorry. Ty Purcell -----Original Message----- From: Derek Listmail Acct [mailto:listmail@oversimplified.com] Sent: Thursday, September 09, 2004 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycon IP 300 SIP vs Grandstream BT-101Deployment > I plugged it in, configured it, and it works great. I

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

I am puzzled by the use of the term "cross-posted" here... I don't see the OP or their question or any similar words from the question involved the the given link, though that link seems worth bringing it to the OP's attention. But the function given in the question seems to have other problems: A) The download.file function call puts its result in a different place than the

--- "Keith O'Brien" <keith@voipreviews.com> wrote: [phone1] type=friend username=keith secret=keith host=dynamic qualify=2000 disallow=g729 auth=md5 context=sip mailbox=9999 callerid="keith@10.1.1.12" <1000> But the log in SJPhone indicates that the registration is being rejected: 2003-09-25 18:55:34.776 UDP LOCAL->10.1.1.12:5060 REGISTER sip:10.1.1.12

Hi Everyone, I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the

Hello, This is cross posted from StackOverflow: https://stackoverflow.com/questions/50314015/is-there-any-method-to-defer-the-execution-of-code-in-r Cross posting is discouraged in r-help. Rui Barradas On 5/13/2018 8:59 AM, akshay kulkarni wrote: > dear members, > > I have created the following function to read a csv file from a given URL: > > function(){ >

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

I recently setup an Asterisk Server- I was able to follow a tutorial from http://www.automated.it/guidetoasterisk.htm#_Toc49248752 Until it told me to call another line, let it ring until voice mail picks up. My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config: