similar to: Asterisk hangs up on incoming PSTN line to analog extension

I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the

The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose.

Hello, I have configured advanced routing with success using three (03) network adapters ( ETH0, ETH1 and ETH2). Now I have to configure an addittion range in the same Interface ETH0 ( another internet ip range address ), so I would like to have TWO different ranges at the same ETHERNET CARD ( in this case ETH0 ) AND having advanced routing running it. I Did the same

hi everyone i have to create a share that will allow a group of users just to create files, not modifying or even reading these files. plus, i need to allow just xml files to be created. is that possible? tks in advance -- *Leonardo de Souza Carneiro* *Veltrac - Tecnologia em Log?stica.* lscarneiro at veltrac.com.br <mailto:lscarneiro at veltrac.com.br> http://www.veltrac.com.br

Hi: We're offering some sort of reward to that one who can help us For this site we are using trixbox and Asterisk 1.2 More info off list. Many thanks, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070927/4b058148/attachment-0001.htm

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hi fellows. I'm new to the list and hope I'll have a nice time here. one of my servers running samba+ldap is not working properly. the setup is: Debian 5 fileserver:/etc/samba#smbd --version Version 3.0.24 fileserver:/etc/samba# slapd -V @(#) $OpenLDAP: slapd 2.3.30 (Oct 5 2008 22:08:58) $

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

Sorry if this question already asked. I'm implementing Voip with asterisk and grandstream gxw4108, according from the manual, for connecting with PSTN I must configure one SIP account and assign that for dialing the PSTN so in my sip.conf I configure SIP account(extension) : [1401] type=friend username=1401 secret=1401 host=dynamic context=my-office insecure=port in my extension.con

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port.

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:

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

Hola Amigos: Gracias por todas las respuestas sobre como filtrar datos en un data frame. Ahora tengo un nuevo problema, necesito instalar R 2.15 manualmente, pero no puedo hacerlo desde un repositorio. Alguna solución??????????' Un Saludos, Leonardo ------------ próxima parte ------------ An embedded and charset-unspecified text was scrubbed... Name: no disponible URL:

Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I

We don't generate any .lib as those don't work well with incremental linking (and give zero advantages when linking AFAIK), and it would be pretty easy to have a modern format for having a .ghash for multiple files, something simple like size prefixed name and then size prefixed ghash blobs. On Fri, Jan 26, 2018 at 8:44 PM, Zachary Turner <zturner at google.com> wrote: > We

Look for this code in lld/coff/pdb.cpp if (Config->DebugGHashes) { ArrayRef<GloballyHashedType> Hashes; std::vector<GloballyHashedType> OwnedHashes; if (Optional<ArrayRef<uint8_t>> DebugH = getDebugH(File)) Hashes = getHashesFromDebugH(*DebugH); else { OwnedHashes = GloballyHashedType::hashTypes(Types); Hashes = OwnedHashes; } In the else block there, add a log

You can make a PDB per lib (consider msvcrtd.pdb which ships with MSVC), but all these per-lib PDBs would have to be merged into a single master PDB at the end, so you still can't avoid that final . In a way, that's similar to the idea behind /DEBUG:FASTLINK (keep the debug info in object files to eliminate the cost of merging types and symbol records) and we know what the problems with

it does. I just had an epiphany: why not just write a .ghash file and have lld read those if they exist for an .obj file? Seem much simpler than trying to wire up a 20 year old file format. I will try to do this, is something like this acceptable for LLD? The cool thing is that I can generate .ghash for .lib or any obj lying around (maybe even for pdb in the future). On Fri, Jan 26, 2018 at

I don’t have pgo numbers. When I build using -flto=thin the link time is significantly faster than msvc /ltcg and runtime is slightly faster, but I haven’t tested on a large variety of different workloads, so YMMV. Link time will definitely be faster though On Sun, Jan 28, 2018 at 2:20 PM Leonardo Santagada <santagada at gmail.com> wrote: > This part is only for objects with /Z7 debug