search for: usercallerid

Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- -------------------------------------------- Paulo Garcia Pika

...X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf --------------------------------------------- ~ [channels] ~ language=fr ~ context=montr?al ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 ---------------------...

...han_zap.c:10876 setup_zap: Ignoring answeronpolarityswitch May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring hanguponpolarityswitch May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring signalling May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring usercallerid -- Reconfigured channel 3, FXS Kewlstart signalling May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring signalling May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring usercallerid -- Reconfigured channel 1, FXO Kewlstart signalling What happen??, Why don...

...be choppy May 28 13:59:42 WARNING[147466]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) [chan_zap.so] => (Zapata Telephony w/PRI) May 28 13:59:42 WARNING[16384]: chan_zap.c:8030 setup_zap: Ignoring usercallerid May 28 13:59:42 WARNING[16384]: chan_zap.c:704 zt_open: Unable to specify channel 1: No such device or address May 28 13:59:42 ERROR[16384]: chan_zap.c:5499 mkintf: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 May 28 13:59:42 ERROR[16384]: chan_zap....

...25-32 fxoks=33-48 defaultzone=us loadzone=us [zapata] [channels] ; context=default resetinterval = never musiconhold=tape switchtype=national context=pri signalling=pri_cpe group=1 echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-2.0 busydetect=no pridialplan=unknown usercallerid=yes callerid=asreceived channel => 1-23 I see the following the full log: Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Executing Dial("SIP/4228-082131e8", "ZAP/G1/1xxxxxx5800") in new stack Oct 4 09:09:30 DEBUG[29894] dsp.c: dsp busy pattern set to 0,0 Oct 4 09:09:30...

Judging from log analysis, there has been in the past some interest in the Asterisk example files that I have made available from my experimental platform (home office) so I've put my newest versions up on my site. Feel free to dig through these and cut pieces up for your own use, or just marvel at the bad logic flows. :) http://www.loligo.com/asterisk/current/ JT

...phone network? My config files are as follows: /etc/zaptel.conf fxsks=1 defaultzone=uk loadzone=uk /etc/asterisk/zapata.conf [channels] signalling=fxs_ks context=incoming channel => 1 busydetect=1 busycount=7 relaxdtmy=yes callwaiting=yes callwaitingcallerid=yes transfer=yes cancallforward=yes usercallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no calleris=asreceived /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer exten => s,2,Dial(Console/dsp) [local] exten => _9X.,1,Dial,Zap/1/${EXTEN:1} Regards, -Jeff Fern

...channels] switchtype=national context=from-pstn signalling=pri_cpe pridialplan=unknown usecallerid=asreceived echocancel=no echocancelwhenbridged=no echotraining=400 overlapdial=yes immediate=no group=0 channel => 1-23 context=from-ctx switchtype=national pridialplan=unknown signalling=pri_net usercallerid=asreceived echocancel=yes busydetect=yes overlapdial=yes immediate=no echocancelwhenbridged=no echotraining=400 group=1 channel => 25-46 Zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Thanks guys!! Dan

...aptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no immediate=no transfer=yes threewaycalling=yes canpark=yes echocancel=yes busydetect=yes signalling=fxo_ks group=1 callerid="ANALOG1" <1001> channel => 1 signalling=fxo_ks group=1 callerid="ANALOG2" <1002> channel => 2 signalling=fxo_ks gr...

...configured as follow: vodka ~ # grep -v "#" /etc/zaptel.conf fxoks=1 fxsks=2-4 loadzone=fr defaultzone=fr (module 1 is one fxs, and 2 3 and 4 are fxo modules, so it is a TDM13B). asterisk is configured as follow: vodka asterisk # more /etc/asterisk/zapata.conf [channels] context=test usercallerid=yes hidecallerid=no immediate=no signalling=fxo_ks echocancel=yes group=1 channel=1 signalling=fxs_ks echocancel=yes group=2 channel=2-4 vodka asterisk # more extensions.conf [test] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Playback(tt-weasels) exten => s,4,Playback(vm...

Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance.

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

...pp:off - - - - a=ptime:20 a=sendrecv m=video 17962 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv Here is my sip.conf -------------------- [general] port = 5060 bindaddr = 0.0.0.0 disallow = all realm = testRealm allow = ulaw allow = h264 videosupport=yes canreinvite = no calleridupdate = info usercallerid = no context = default [toTandberg] host=192.168.50.53 type=friend qualify=yes qualifyreq=1 ------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100809/86f992c3/attachment.htm

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

...a flash hook. The call gets disconnected when the flash button is pressed. I have transfer=yes, threewaycalling=yes etc enabled in my zapata.conf . signalling=fxo_ks threewaycalling=yes transfer=yes callwaiting=yes flash=1000 context=from-internal group=1 callgroup=1 pickupgroup=1 hidecallerid=no usercallerid=yes musiconhold=default channel => 97 Here are the asterisk debugs that I get when I flash the call ------------------------------------------------------------------------------------- Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Exception on 106, channel 97 Oct 25 10:27:13 DEBUG[28965] chan_zap....

Due to some failed hardware on another platform, I've had to move a T100P card to a different chassis. After this move was completed, I am seeing some strange results on the T100P card that do not display to me any failure mode with which I am familiar. The card comes up, and shows "good" carrier and status, but Asterisk refuses to bring the D-channel up. Calls of course do

...13 Zaptel: 1.4.6 Libpri: 1.4.2 Configuration of zaptel.conf: loadzone=de defaultzone=de span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 channels = 1 zapata.conf: [trunkgroups] [channels] overlapdial = yes language = de switchtype = euroisdn pridialplan = unknown prilocaldialplan = unknown usercallerid = no hidecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = no threewaycalling = no transfer = yes cancallforward = yes callreturn = yes echocancel = yes echocancelwhenbridged = no resetinterval = never rxgain = 0.0 txgain = 0.0 group = 1 callgroup = 1 pickupgroup = 1 immed...