similar to: AEAP experience

Displaying 20 results from an estimated 20000 matches similar to: "AEAP experience"

2005 Sep 23
0
Problem with outbound calls
Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>
2005 Sep 12
1
optimizing for via C3
Hi I'm trying to build an Asterisk packages for a C3 system (256MB memory, cpuinfo below). /proc/cpuinfo: processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping : 8 cpu MHz : 1000.736 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu
2006 Apr 24
1
E1 testing
Skipped content of type multipart/alternative-------------- next part -------------- Console logs from Asterisk A: Executing Dial("SIP/test0-5821", "Zap/6/327557670||Tt") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 6/327557670 -- Zap/6-1 is proceeding passing it to SIP/test0-5821 -- Accepting UNAUTHENTICATED call from 195.66.73.122:
2020 Jun 22
2
Voice broken during calls (again...)
Hello, there is no need to change canreinvite for provider configuration. Do not change MTU. Probably there will be another problem. I expect packet size 1466 would pass and higher will have the same result. It would be interesting to make the same test from the outside towards your asterisk with size 2 bytes larger the highest you are able to ping. Marek 2020-06-22 22:26 GMT+02:00, Luca
2020 Jun 22
0
Voice broken during calls (again...)
Am 22.06.2020 um 22:12 schrieb Marek Greško: Hi Marek > Would you mind repeating the test with canreinvite=no set for all you > phones and mobile phones? All my peers have already canreinvite=no... I only have canreinvite=yes on the SIP configuration on the Telekom part: [pbxluca] type=peer defaultuser=111111111 at t-online.de secret= xxxxxxxxxx dtmfmode=rfc2833 host=tel.t-online.de
2006 Nov 23
1
asterisk 1.4 chan_h323, help please...
Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best
2004 Feb 03
1
Problems with chan_sip: random calls have no sound withouth any errors
Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all
2006 Nov 05
1
asterisk DTMF detection
Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and
2005 Feb 15
0
Re: Asterisk-Users Digest, Vol 7, Issue 216
asterisk-users-request@lists.digium.com is believed to have said: >Hey Everyone, > >I downloaded and installed the X-Lite softphone the other day (the lite >version) and cannot seem to get it to work well. > >Don't get me wrong, it registers with my asterisk server and everything >seems to work well, except the call quality really is horrible. > >I thought it may
2024 Jan 25
0
asterisk release 18.21.0
The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2024 Jan 25
0
asterisk release 18.21.0
The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2024 Jan 25
0
asterisk release 20.6.0
The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2024 Jan 25
0
asterisk release 20.6.0
The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2024 Jan 25
0
asterisk release 21.1.0
The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2024 Jan 25
0
asterisk release 21.1.0
The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!
2009 Jan 30
2
Asterisk with Avaya
Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --
2006 May 10
0
No audio in either direction on Zap -> SIP or SIP -> Zap calls
Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and
2006 Feb 14
1
fax pass-through
hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35
2020 Oct 21
0
Asterisk 18.0.0 Now Available
Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several
2020 Jan 21
2
astricon videos
Hello, any plans for astricon videos? Thanks Marek