similar to: problem to install asterisk on vps digitalocean

Dear All, Is there anyone who is having FastAGI support for C & C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. Thanks in Advance, Kashyap -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi list , is having problems when sending a fax with hylafax and a card sangoma A200D, when he sends it arrives to the destination but it paginates appears in white this is my log Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms, 20 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM WWW.SOFT-SWITCH.ORG spandsp/ Aug 20

Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com

I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking

2015-03-18 11:13 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > Hi , I'm trying to apply this patch from the source asterisk > asterisk-11.16.0 and when I apply it shows me this message > > asterisk-11.16.0]#patch -p0 < refs > patch: **** Only garbage was found in the patch input. > > is the correct way to apply the patch or am I doing wrong? >

Hi list, I install dahdi-linux successfully with the module of oslec for the echo, but when I specify it in the system.conf the echo canceller oslec it shows me errors: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) I see that the echo cancellers is supported: mg2, kb1, sec2, and sec because oslec is not supported?, but he has support to compile it with dahdi_linux! best

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com

Hi, we have a faxserver with Asterisk, IAXModem and Hylafax. Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax. Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk. We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user. Sometimes Hylafax reports that a modem

Hi I have a problem w/ voicemail, the subject message is corruption when used voicemail variables, e.g. : voicemail.conf emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} Return: Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= Expected: Subject: 1504|12|"Teste - Rafael" <1570>|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51)

2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the

Starting with 3.7 rc1, my workstation seems to loose ram. Up until (and including) 3.6, used-(buffers+cached) was roughly the same as sum(rss) (taking shared into account). Now there is an approx 6G gap. When the box first starts, it is clearly less swappy than with <= 3.6; I can''t tell whether that is related. The reduced swappiness persists. It seems to get worse when I update

Cheers all, you may have seen related preparatory news on the NUT website or in the README changes, but now the next step is official: The fine folks at DigitalOcean approved FOSS sponsoring credits to re-host the custom-built multi-platform non-regression NUT CI farm, which builds several hundred scenarios per iteration (to cover a matrix with many OSes, toolkit versions and

Cheers all, you may have seen related preparatory news on the NUT website or in the README changes, but now the next step is official: The fine folks at DigitalOcean approved FOSS sponsoring credits to re-host the custom-built multi-platform non-regression NUT CI farm, which builds several hundred scenarios per iteration (to cover a matrix with many OSes, toolkit versions and

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

Hi all,I hope to find a solution with the help of the list, I'm trying to get the status of my extensions with ejabberd , the idea is to visualize my users ejabberd incoming calls or missed. I'm testing with my operator extension with this code but only get the missed call notification does not show me where the call is coming. my piece of code [operadora] exten =>

Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX, generate tables in a couple of files in the folder realtime / mysql , voicemail_messages.sql and voicemail.sql the connection with mysql and odbc works well isql asterisk useradmin xxx +---------------------------------------+ | Connected! | | | |

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next