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Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello people, I've ran into two problem that I can't seem to be able to solve on my own. Here's my scenario (running Asterisk 13.28.1): In short: - Asterisk behaves unexpectedly (at least to me) when negotiating between endpoints             that have a different but intersecting set of codecs (preventing direct media flow).           - Also, when an endpoint sends RTP with an

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom

Stack: (gdb) bt #0 0x0000000000aaf38a in silk_VAD_GetNoiseLevels (pX=pX at entry=0x7f26740297a0, psSilk_VAD=psSilk_VAD at entry=0x15897c38) at silk/VAD.c:315 #1 0x0000000000aa4a9d in silk_VAD_GetSA_Q8_sse4_1 (psEncC=0x15897c18, pIn=<optimized out>) at silk/x86/VAD_sse.c:177 #2 0x0000000000a9f92b in silk_encode_do_VAD_FLP (psEnc=psEnc at entry=0x15897c18) at

Hi Jean-Marc, gdb out is "Program terminated with signal 8, Arithmetic exception." most likely this division by zero. you're right, this crash is reproduce on seq number 4294967265 (20ms rtp packet). This is about 994 days. "Jean-Marc Valin" <jmvalin at jmvalin.ca> писал(а):Hi Dmitry, > >So it's not explicitly in your report, but it looks like the crash

Hi, I use SpeexEcho in my own softphone using MGCP protocol register to a PBX under WindowsXP in my laptop. I make a call: softphone ---> PBX ----> MediaGateWay ---PotsLine---> Traditional Phone. In the other side(TraditionalPhone), I did not here any echo voice, SpeexEcho works well. But, if I let laptop to make some sounds, such as play music, I heard all

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I want to replace a Telebutler software auto attendent system that used a 4 port Dialogic board connected to a Panasonic KXTD 1232 6 line system. We have spare computers here. How do I connect asterisk to this Panasonic system? -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI, Does anyone know if there is a PCI-X 4 port PRI cards available on the market? If so, have anyone used it and how reliable they were? Any help is appreciated...

I've tested a few different wifi SIP phones for office/factory use, and generally have been underwhelmed. Before I grab another few and test, I'd like to ask around here about the candidates. My requirements are relatively simple: - WEP/PSK should be supported WITHOUT dragging the phone down - roaming between access points without dropping the call - decent set of ringers, not the

Hi Dmitry, So it's not explicitly in your report, but it looks like the crash is due to a divide-by-zero at: min_coef = silk_DIV32_16( silk_int16_MAX, silk_RSHIFT( psSilk_VAD->counter, 4 ) + 1 ); which happens because counter is -16 (which means (-16 >> 4) + 1 == 0). Now, this could be caused by an integer wrap-around, but it should only happen after encoding around

Hello We have around 50 phones in our company, and I am playing with the thought to gradually go over to using sip services and ip-phones internally. However at first I would liked the Asterisk just to sit between the phone line and the Panaosnic, so I can take out one lin/number at a time to use ip phones. I am new to Asterisk, and haven't done much configuring of the PBX either. So I also

Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? Mike. -------------- next part

...canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card features: * Supplementary Services o Number Identification services (CLIP, CLIR, COLP, COLR, KEY, MSN, DDI, SUB) o Call offering services (TP, CFU, CFB, CFNR) o Call completion services (CW, HOLD, ECT)...

Hi, I am starting to look at the costs involved in putting a PBX system together using *.. We will be using ISDN lines in the UK, which when installed provide a box on the wall with 2 analog ports and 2 digital ports, so these seem to be the options available to me.. 1. I could use an ISDN board but I have not really read good things on the mailing list about using ISDN boards, and I suspect

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more