similar to: Zaptel/Zapata and SIP relationship

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

i've been using *1.2 w/ realtime static zapata in mysql table fine. but after i upgraded to 1.4. it seems like the zapata table doesn't load correctly. i have to go in the console and use the "zap restart" to get the zap channels register. is this sounds like a bug or something i'm missing when upgrading to 1.4? -- Edwin Lam <edwin.lam at officegeneral.com> Systems

is where anyone who can tell me how it's possible to set nationalprefix & internationalprefix for a single isdn-card and not for all installed cards ?

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...

I just installed two new X100P's today in my RH7.3 box and was poking around /proc What does the RED refer to on board 2? /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 /proc/zaptel/2 Span 2: WCFXO/1 "Wildcard X101P Board 2" RED 2 WCFXO/1/0 Any clues on how to setup channels on the two interfaces?? I am up to here in

hi! when i have busy signal on analog line (zap card) it doesn't detect that line is busy ? is it possible to change detected sequence (frequency) of busy tones on line (zapata.conf ??) tnx, Thomas my zapata.conf [channels] language=en context=lin1 signalling=fxs_ks channel => 1 group=1 echocancel=yes echocancelwhenbridged=yes rxgain=3.0 txgain=3.0 busydetect=yes

Hi, Do you have some experience with the "best" values for those parameters in youyr particular case? I mean the best raport between sound level in both direction and echo cancellation. For me, the best result I can get is with: rxgain=10 txgain=15 ... the sound level is good, but the echo is a little bit to strong for my taste. Something interesting is that if I put a txgain value

Hi I've problem, i cant get tone on a FXS ZAP channel my configuration are: -- zaptel.conf -- fxoks=1 --zapata.conf -- [channels] immediate=yes context=bell signalling=fxo_ks channel=1 --extensions.conf -- [home] exten => 500,1,Dial(IAX2/guest@misery.digium.com/s@default) [bell] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,Dial(${PHONE},16,tr) ANY IDEA?

Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface

I'm not sure how I can handle timeouts and invalid extensions for my Zaptel channels... Their default context is [internal], and in internal I have defined extensions i and t to handle timeouts and invalid extensions. However, the default for the Zaptel channels is "immediate=no", so the default context is only "run" when input has been entered. This means that the i

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on