similar to: Internal Data Stream Error

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

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

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

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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

G'day list, It appears I got ahead of myself in believing I'd solved my CND (caller id) problem. My callerid.c change got it working on one of the handsets I have, but not the other. (Recap: The AU CND standard has an increased delay between the end of the ring signal and the start of the CND FSK data -- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030). I've

Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz On 11/14/17 5:06 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: >> On 11/14/17 4:27 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >>>> On 11/14/17 3:55

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everything

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

On 11/15/17 11:10 AM, Joshua Colp wrote: > On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: >> On 11/14/17 5:23 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: >>>> Trace with 3 clients. We can hear each other but no video. >>>> >>>>

On 2/22/18 3:46 PM, Antony Stone wrote: > On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: > >> On 2/22/18 1:07 PM, Antony Stone wrote: >>> On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: >>>> Usually phone companies set the outgoing CallerID for you but >>>> >>>> recently we got control over that and are

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

Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the

On 2/22/18 1:07 PM, Antony Stone wrote: > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: > >> Usually phone companies set the outgoing CallerID for you but >> recently we got control over that and are now setting the outgoing >> Calleir ID ourselves. My problem now is that the CDR will put the >> outgoing CID in the CDR instead of the extension

Hi, I have a problem for install chan_capi My pc: Suse 9.1, with asterisk current stable en cvs I have download http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz And the path from http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 And patch the chan_capi # cd chan_capi-0.3.5 # patch -p1 < chan_capi-0.3.5-patch.diff # make /for install My error

Hello I've never built an IVR before, so I was wondering if someone could share some code from their extensions.conf that would perform some of thoses steps: 1. When a call comes in from the TDM FXO port, answer 2. If no CID, play message "No CID available. Please type the number where you wish to be called back". Loop until OK or remote party hung up 3. When CID is available,

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has