similar to: call fail from audiocode to sip trunk

Displaying 20 results from an estimated 200 matches similar to: "call fail from audiocode to sip trunk"

2010 Sep 18
2
Audiocode Median 2000 Gateway with Asterisk ?
Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I
2007 Jun 21
2
mediant 2000 with asterik configuration
Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2010 Apr 11
0
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode
Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr
2007 Sep 06
2
FAX machine connect with audiocode SIP device
Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device --------------------------------- Ready
2007 Aug 08
1
asterisk wait for traling digits
Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when
2010 Apr 10
1
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant
2015 Sep 25
2
Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)
Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>: > Hello! > ? > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or instructions on > how to proceed? > ? > Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?
2004 Sep 16
2
Audiocodes Mediant 2000
Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar
2007 Aug 22
0
asterisk with FAX problem
Dear all I have setup of asterisk 1.2.14 and this is working fine. first i want to explain you my setup of asterisk on network i have connect my asterisk with mediant 2000 gateway and PRI terminated on mediant. [fax_machin]------[audio_code_fxs]-----[Asterisk]-------[mediant_2000]---PRI--<--< my fax machine connected with audiocode 24 fxs extention and which is
2007 Jan 15
2
Audiocodes Mediant 1000, Polycom, and no ringback on transfer
I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any
2014 Feb 11
1
file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory
Dear Folks, [Test_Context] exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _911.,2,Set(CALLERID(num)=xxxxxxx) exten => _911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)}) exten => _911.,5,Set(${CALLERID}=${CALLERID(num)}) exten =>
2003 Nov 23
2
SIP Express Router & Asterisk
Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or
2010 Feb 25
2
Do i need install Dahdi or libpri ?
hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>
2006 Oct 14
0
SIP trunk from an Audiocodes mediant 1000
Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten =>
2007 Jun 25
2
Rining 180 and 183
Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of
2009 May 15
0
Mediant 1000 audiocodes and Trixbox
Hi, This is my first experience with a mediant 1000 and an Asterisk Trixbox. the mediant has 12 FXOs and 12 FXSs, and I want to use it them all. I will have extensions connected to the FXS ports, and lines to the FXO. Can anyone guide me, please? regards, -- Guillermo Garron "Linux IS user friendly... It's just selective about who its friends are." (Using Ubuntu, Debian,
2007 Jan 16
0
Asterisk 1.2.14 and Audiocodes Mediant 1000
I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the
2001 Sep 18
3
Vorbis as a Quicktime codec
Anyone wants to create one? Below is a link of a sample audio codec, maybe it helps. http://developer.apple.com/samplecode/Sample_Code/QuickTime/Codecs/audiocode c.htm _________________________________________________________ Do You Yahoo!? Get your free @yahoo.com address at http://mail.yahoo.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:
2003 Jun 02
0
SIP, DTMF, and AudioCodes Mediant 2k
Greetings... I'm working on getting an AudioCodes Mediant 2000 big box o' PRI's going with Asterisk, and am running into a problem with DTMF handling. The box is sending DTMF packets to Asterisk as INFO packets, and they are apparently being seen by Asterisk. However, the DTMF knowledge doesn't seem to actually do anything -- the VM system doesn't recognize the digits,
2007 Apr 12
2
Best External PRI Gateway?
I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are