similar to: Peer to Peer calls

Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Thanks Andrew. I have tried with recent tblgen, ReadAdvance would not work for multiple latencies. Maybe I should make improvement into tblgen if Pierre-Andre does not have the change anymore. However, I just a little curious about the situation I met. The hardware forwording may fail for different reasons, which different register read may have different latencies, depending both on the register

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

It does not work. I have tried to use the latest master today. But tblgen still give me information like error: Resources are defined for both SchedRead and its alias on processor MyArchModel def : ReadAdvance<MyReadVector, 3, [MyWriteAddVector]>; ^ Unless I change "MyReadVector" to another read like "MyReadVector1", it would not work. Debugging into tblgen, there is

Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > > Hello, > > > > I have an Asterisk 15 with PJSIP behind

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward

Dear sir or madam I want to know how to use "metafor"package do a meta anlysis, forest plot show the results of whole study and results of two subgroups like this the forest plot show in the attachment. looking forword to your reply Regards and Cheers Xudong Liu School of Public Health and Primary Care The Chinese University of Hong Kong Email: liuxudong at cuhk.edu.hk

I am terribly sorry to bother the list with such generic and bizarre problems, but I have been racking my brain with these for the last week working on it for at least 60 hours. If anyone can even point me in the right direction I would be eternally grateful. So without further adu here are my woes: I have * (2004-04-09 CVS) running on a P4 1.6Ghz CPU, 512MB RAM, Debian "Sarge", and

For my opinion, special function not needed, because at first time, oai set to NULL, freeaddrinfo tried to free each ai, until occurance of ai_next == NULL. But it works. Linux, Windows, coordinator with white IP, other nodes behind NAT's (1 or two NAT's, only direct connections allowed). There are some problems with MinGW make, at least - mingw-64 from Fedora 25. 1. No definition for

Hi, I'm using the ITU-T P.834 methodology to derive the Ie parameter for Speex. According to the E-model, Ie indicates the degradation in quality of a codec at 0% packet loss. It is a non-negative value: the higher, the worst. Details in ITU-T G.107. Because Speex is multi-rate, we are deriving one Ie parameter for each Speex rate. In our preliminary experiments, we've got a value of

Hi list, I happened to read below thread (written in 3 years ago). I think I may need this ReadAdvance feature to work with my ARCH. It is about the scheduler info which describes reading my ARCH's vector register. There are different latencies since forwarding/bypass appears. I give it as below example: def : WriteRes<WriteVector, [MyArchVALU]> { let Latency = 6; } ... def

Good afternoon! Module - net_socket.c Function - get_known_addresses --------------------------------------------------- struct addrinfo *nai = xzalloc(sizeof *nai); if(ai) ai->ai_next = nai; ai = nai; -------------------------------------------------- For my opinion, possible causes: 1. Lost trails (ai_next) 2. ai_next not initialized 3. Possible segfault during

I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As

I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and