search for: zap1

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'zap1' number is '4521' dialparties.agi: Methodology of ring is 'none' &gt...

R-help, Given a simple linear model, say lm(x ~ y + z), I would like to remove model terms that are factors with only one level. Thus, if 'z' were a factor with only one level, lm(x ~ y + z) becomes lm(x ~ y + 1). Likewise, if both 'y' and 'z' are one-level factors, then the resulting calculation is simply lm(x ~ 1). Unfortunately, I have not been able to come up with an

...waitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancelwhenbridged=no echocancel=64 echotraining=800 callgroup=1 ; i do not use call groups, left it in as it's default pickupgroup=1 rxgain=0 txgain=0 group=1 immediate=no context=from-analog-zap1 faxdetect=both ; normally would be: none channel => 1-2 ; normaly would be: 1 ;channel 2 ; normally would be uncommented ;echocancelwhenbridged=no ;echocancel=64 ;echotraining=800 ;rxgain=0 ;txgain=0 ;context=from-analog-zap2 ;faxdetect=both ;immediate=no ;group=1 ;usecallerid=no ;signalling=fx...

...s in the following string when i run /etc/init.d/zaptel restart: Jan 1 10:48:16 bu kernel: usbcore: deregistering driver wcusb Jan 1 10:48:16 bu kernel: Freed a Wildcard Jan 1 10:48:16 bu kernel: Zapata Telephony Interface Unloaded Jan 1 10:48:16 bu udev[27206]: removing device node '/udev/zap1' Jan 1 10:48:17 bu udev[27207]: removing device node '/udev/zaptimer' Jan 1 10:48:17 bu udev[27222]: removing device node '/udev/zapchannel' Jan 1 10:48:17 bu udev[27233]: removing device node '/udev/zappseudo' Jan 1 10:48:17 bu udev[27244]: removing device node '...

...lace. Any ideas? Error Opening channel:2 not available, see va_g729_init_global(..) WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b resource for channel 2 WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 format Hangup Zap/1-1 2) have discovered a problem when using g729 under the following setup: SIP call between a Budgetone 102 and ATA 186 (configured without silence suppresion). Both ends...

...:9155 setup_zap: Unable to register channel '1' Mar 13 20:43:35 WARNING[5779]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 ] As a matter of fact this is what is listed in /dev/: aaberga@epsilon-stargate:~/sources/voip/zaptel-1.0.6> ls /dev [....] z2ram zap1 zap2 zap3 zap4 zapchannel zapctl zappseudo zaptimer zero zkshim zqft0 I edited the list to avoid a huge message. As said I am not a Linux low level expert ... still it's striking that Asterisk does not find a pesudo file /zap/channel and there is something similar, ie zapchannel. Anyways the...

Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling...

...ne. the problem is when the caller hangs up the asterisk does not. after caller hangs up and tries again he always gets a busy tone until asterisk stops by timeout. i get this meesage in asterisk message is too long, ending it now... timeour zap/1-1 spawn extension(tumpak,t,1) exited non-zero on Zap1-1 earlier i tried it without the exten=>t,1,hangup line and i got no rule 't' in tumpak.. and added that . but i still get the same thing. can anyone help me with this. thanks cm ===== Designs __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mai...

...aded zaptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file is /etc/zaptel.conf line 206: Unable to open master device '/dev/zap/ctl' I checked /dev/ and there is no /zap directory but, instead, some devices that look relevant like zap1, zap2 etc. Obviously, if boot with the older kernel, wctdm will not load either. Any help is welcome Thanks in advance, dionisis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/c688e9ea/attachme...

...any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf ----------- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=????? (not the real one) host=dynamic mailbox=1000 canreinvite...

...39; properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug- devel@lists.sourceforge.net> Nov 24 10:23:44 jfd wait_for_sysfs[3377]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug- devel@lists.sourceforge.net> I added the udev rules as described in README.udev. Any ideas? TIA -------------- next part -------------- An HTML attachment...

...my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However, after my mobile hang up, the Trunk Zap1 does not. I've to reboot the computer to free up the line and it is also not possible to do a graceful reboot because I would get a kernel panic. I'm actually using PSTN for the trunk. Hope anyone can provide me some advice, could also be a link to another post which I might have missed...

...have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same context. I have two X100P (Zap1 & Zap2) and one S100U (Zap3). If I use my S100U and dial extension 800, it works. It calls. However when I copy my 1.call file. it says: Unable to create channel of type 'Zap'. Does anyone have any suggestions? or know what am I missing? T...

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

I have 2 sip phones and an analog phone attached to a Digium USB fxs device. I would like the analog phone to ring when transfers are made to it, but I don't want it to ring when a call comes in from outside, although I would like the person at that phone to be able to pick up the phone and answer the incoming call. Is that possible? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology

Can someone please tell me if this is how call waiting should work? ZAP/1 and ZAP/2 are X100P on phone lines ZAP/3,4,5 have phones conected. If a call is in progress between ZAP/2 and ZAP/3 and ZAP/2 gets a callwaiting signal which ZAP/3 hears the Dial command has SIP/2002&ZAP/3&ZAP/4&ZAP/5 If ZAP/4 is off hook, I hear the call waiting signal The CLI says it is ring the 4

...e G3, can asterisk determine that the call is coming in from 5611, > and will it understand the G3's caller ID? > Only if you can configure the G3 to pass caller id info, our G3 admin couldn't. the 4 lines you want to pass to Asterisk will connect via their own ZAP channel. 5610=ZAP1, 5611, ZAP2 and so on. You can use that in your dial-plan to pass incoming to certain extensions on Asterisk. Since this is only a proof of concept, this should do very well. But, keep in mind that the X100P cards have limits on the number of cards within 1 PC. They generate LARGE amounts of...

...his? Do I do ; [Nikotel] type = peer username = fred secret = blah host = nikotel.com or host = calamar0.nikotel.com ? and how does this relate to the above "register" business, why specify this twice? Finally, how do I dial? At present, if I dial 9, it goes straight out on channel Zap1-1 (FXO card). What is the best practise for alternative routes? Should I have it dial via Nikotel if one dials 8 instead of 9 for an outside line? Again, if I wanted to do this, I can't quite see from the examples and the explanation in the handbook how I would do this. I guess the example...

AS a proof of concept experiment, I want to try and integrate Asterisk with my Lucent Definity G3 switch. I don?t have an available T1 port on the G3 but I can round up 4 analog ports off the G3. What I thought I could do is create a ?Hunt group? on the G3. Let?s say I configure 5610 ? 5613 as a hunt group. I know I would need a 4 port FXO card to install in the Asterisk server. Two configuration

...error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The outgoing call dials out on Zap2 and is received by SpanDSP on Zap1. I'm not too sure what the Zaptel device is using as a timing source or how to alter it in order to tune the carrier frequency. If anyone has any ideas I'd be very appreciative of your assistance. Many thanks, Stephen. -- Starting simple switch on 'Zap/3-1' -- Executing...