similar to: Anyone have experience with chan-capi in Australia?

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612

Hi, I'm after some ISDN card resellers in Australia. I've noticed the AVM cards should work well here, however I've found the reseller here has no website, is listed in Yellow pages as management consultants (??) and when I called quoted almost $400. Does anyone know where I can get the AVMs in Australia a bit cheaper or another card (not the Netjet) that works with Asterisk.

Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL

Hi, Just wondering if there are any people out there using the Zaptel Wildcard E1 interfaces on Telstra PRI?s in Australia. If so, is anyone happy to share their config? Thanks, Phil --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.510 / Virus Database: 307 - Release Date: 14/08/2003

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

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

Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes

Hello list, I have an X100P clone card, and I've been trying to get it to receive Telstra Caller-ID without any luck. Hence these two questions:- 1) Will a genuine X100P work? 2) If so, where can I get one in Australia? Cheers, Jon

Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Hi does anyone have a definitive list of countries other than those listed on the wiki which are supporting app_SMS on landlines using ETSI ES 201 912 ?? Thanks Tim Robinson Basingstoke, UK

> > That would be called a milliwatt generator. It likely exists in their > central office, but its typically used by their technicians to ensure > new installations meet specs and sometimes in troubleshooting. Call your > telco repair number and see if they will give you the telephone number > for it. I'll try that. 1st line technical support at Telstra are not known for

On 11/22/05, Sharon Kimble <sharon04 at gmail.com> wrote: > Does anybody know the gnome program that you can write in latex with please? > > I'm not sure but I think its in the base repo. Its not winefish as it > (winefish) doesn't compile the source text like the program that I'm looking > > for does. > > Hopefully thanks > Sharon. > -- I am

I have an x100p which suffers from echo (no surprise there apparantly :), and a few of the things I've read about tuning out echo say the first thing to get right is the tx and rx gain, and for that you need a few different types of test lines from the telco (Telstra, Optus, whatever). Does anyone have any numbers in Australia for such lines? I think they just need to send out a tone of a

Show this message adduser: Warning: The home dir you specified already exists adduser: The user 'asterisk' already exist, and is non a system user. dpkg: errrore processando asterisk-cah-capi (--configure): il sottoprocesso post-installation script ha restituito un codice di errore 1

When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process'

Hello all, I've just finished to install chan_capi with 3 AVM Fritz PCI cards. It correctly loads the 3 drivers, and * starts without errors. immediately after * start, audio quality is really fine, but, after the first incoming call, all incoming audio is broken, trembling and stuttering.

Hi, i have a debian-system with the asterisk 1:0.9-1 packages and chan-capi 0.3.1-2 installed. My chan-capi seems to be out-of-order. Capi and I4l work in general. I can use isdnlog and capi4hylafax. Using chan_modem_i4l with the sam context work fine too. But no incoming calls are answered by asterisk over chan-capi. If have 2 isdn-card: a passive avm a1 (i4l/hisax) and a active avm b1

Hello List and Armin, I have been trying to narrow down my problems with getting chan_capi to function properly. It seems that anything above build 8015 causes a segfault on dial or receive. The problem almost seems sporadic, and is certainly related to sip or iax channels. As soon as I update to build 8016, the problem starts. 8015 is fine. For example, if I direct the number right to a

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue?