similar to: Message Indicator Light

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

Let post this question.. Because I must be real slow... The following is my config on this... group=1 context=default signalling=fxs_ks channel => 1 context=local signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hi there, I'm very new here and would like to know if anyone has reccomendations on fundamental reading (other than the handbook) whick might prevent me from asking some really dumb questions (after this one of course). What I'm trying to do: I have a SOHO... very SO in fact. I would like to build a simple system with two analog lines. I do recording and can't always have a cell

Hello, I am trying to use 2 netjet cards under asterisk and isdn4linux. I am having a hard time trying to get them to work in terms of dial out. Does anyone have a working config I could look at for even one card (tried that, not much luck either). When i dial out: -- Accepting AUTHENTICATED call from 172.16.11.2, requested format = 2, actual format = 2 -- Executing

Is there anyway for a sip station to play MoH out of the speaker? I know I can do it by calling the station and putting it on hold. For example: On a samsung pbx with MoH, if you goto one of the workstaions and press a button The moh plays out of the speaker. Is there any way to do this with asterisk? Kevin, Honeycomb Internet Services

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Hello, I faithfully followed the instructions from: http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk but still the message waiting indicator doesn't flash when a message is waiting. There is a brief intermittent chirp but nothing more. Using latest firmware 1.4.1 Thanks for your suggestions,

Hello, I have a problem with asterisk when dial out to a SIP provider. Asterisk send a INVITE with no credentials, the provider reply with a 401 Unauthorized. However, Asterisk DOES NOT resend the invite again with credentials. But it hangs there (maybe waiting for a ok) It is this a bug in asterisk or the provider is supposed to send something else rather than a 401 as answer for a INVITE ?

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David -- David J. Sussman, MBA email:

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris