similar to: X100P callerid ETSI - caller*ID failed checksum

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel

All I would be interested if the current cvs zaptel, has fixed any callerid check sum fails on x100p's. Mark worked on an issue i had with a false 2nd ring detection from the fxo ports on a CAC Adit 600 interfaced to a T400. He added logic to the zaptel driver to ignore a 2nd ring detection if it comes to close in ms to the first ring detection. A false 2nd ring detection could cause the fsk

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Hi, I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines. With only one of those, CID is shown. Beside that, everything is working OK. Lines have different providers and/or locations. All are located in France and CID Detection Method is ETSI FSK / Bell 202. If I'm connecting a TDM400-enabled Asterisk system, to one of those 2 non-working lines : it does work. The only

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

hi: in our country callerid is sent with fsk. but it will sent DTAS(dual tone alerting signal) first, then fsk callerid, then first ring. I search google, but didn't find the configuration method or patch for this. any good suggestion for asterisk to detect this? I have trid asterisk 1.4 and 1.6.2. thanks for help!! Regards, tbskyd

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Has anyone seen the following problem? Until recently, I couldn't understand why some extensions on my * system would have a "congestion tone" as soon as I picked up the handset. A little sleuthing through the logs and the source code led me to understand that * thought it had seen the extension go off-hook, send some DTMF tones, and then wait. * treated this situation as a

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

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hello, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Samba-LDAP PDC at an office and 5 BDC's at other offices. At corporate HQ I have a W2k Server and domain. I have properly configured an interdomain trust and Users in the Samba domain can get to sections on the W2k machine regardless of location. However, members in the W2K domain can only access shares on the PDC. Attempts to access shares on a BDC cause a user name

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

At the risk of being flamed.... Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (?20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is