similar to: dialplan help!-RESOLVED

Hello, I have the 1.0.5 and 1.1.12 versions of Speex. The problem is that when I compile Speex using MinGW and MSys using: ./configure --enable-shared --prefix=/c/bgw/ I don't get neither a DLL file nor a .a library although the compile is successful. Also I must notice that the headers for Speex are installed correctly in the /c/bgw/ directory as specified by the prefix, and I did

I guess it is possible to have an HFC-S card and a Digium TE110P card working at the same time? The TE110P will work in E1 mode. I think the zaptel.conf is probably right but the zapata.conf not (I just tacked on another group at the end but I don't really know what I am doing). Can anyone help? ----------------------------------------- My zaptel.conf looks like this: # hfc-s pci a span

I am having problems loading the zaphfc from bristuff and wcte11xp drivers at the same time. If I load zaphfc then all works fine. If I then load wcte11xp, the card using the zaphfc doesn't pick up calls anymore. I am using bristuff 0.2.0-RC5. Anyone else seen this problem, know of a fix, or can tell me what I am doing wrong? Thanks My zaptel.conf: # hfc-s pci a span definition # most of

I've got a telecommuter working out of her home office, using a Snom 200 phone, what happens is occassionally her phone will loose audio one way. She will be talking on a call that was incoming to her extension, and all of a sudden the caller can not hear her any longer, she can here the caller fine, this has happened with both Sip->Sip calls, and calls that have come in over our PSTN

Bogdan Mustiata wrote: > Hello, > > I have the 1.0.5 and 1.1.12 versions of Speex. The problem is that when > I compile Speex using MinGW and MSys using: > > ./configure --enable-shared --prefix=/c/bgw/ > > I don't get neither a DLL file nor a .a library although the compile is > successful. <snip> > Am I missing something trivial? Maybe not.

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Hi I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.....I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. Anything suggestions as to what I could try