similar to: ISDN CAPI and anonymous callers

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

hi I have a passive isdn port configured in modem.conf in extention.conf i use this two channel (ttyI0 and ttyI1) with the string: exten => _NXXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r how can i hide my msn? is it possible to activate the clir with the @ before the ${EXTEN}? thanks Cristian

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Hi There, i want to stream some pictures out of my application using the Theora codec. I can stream to a socket or a file. The current work based on the png2theora example. Something while enconding went wrong: I can see the whole video in VLC, but the complete duration of the file will not be displayed. If i play the video in Firefox, the first 2 seconds will not be played. I think the

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

Hello! I just encountered a strange thing in my mysql cdr records. From a certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR flieds in the cdr table. As far as I know no changes happened to the system on that date and until then CLIR are recorded properly. The CLIR is still transmitted by the PRI and is shown in the console when a call comes in. But no traces of it in the CDR.

Hi everyone, Im having some troubles getting work Asterisk as SIP Client and a Huawei softswitch as SIP server. I already got my asterisk registered to the Huawei. Im working with a Sipura SPA 2000 registered to Asterisk. When im trying to make an incoming call from the Huawei to asterisk it rings but when i answered the call drp down inmediatly. The sip debug finally show this

When a user calling over the PSTN network calls one of our SIP users with a restricted number (CLIR), our PSTN gateway is sending us incoming calls with the following additional headers: Proxy-Require: privacy Anonymity: uri Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=uri as opposed to Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=off when CLIP

Hi, i got a server that is massivly contacted on the ports 5190, 5150 and 6667. Is there any possibility to stop these requests ? I block everything with my firewall. That are ten thousands of different IP's that try to connect. Thanks and best regards Niels -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://bugs.freedesktop.org/show_bug.cgi?id=104181 Bug ID: 104181 Summary: [NV50/G94] system freeze on youtube with epiphany 3.26 Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

Does rsync work over an NFS mount? --Karl _________________________________________________________________ Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp

Hi All, Server: Fedora 26 samba-4.6.8-0.fc26.x86_64 Workstations (5 of them): XP Pro SP3 I set all five of my customer XP workstations to Send NTLMv2 response only\\refuse LM and NTLM and turned off (smb.conf) lanman auth = yes ntlm auth = yes And had to turn it right back on as the customer's Xerox Workcentre 3550 multifunction printer scanner requires it What are

I am trying to rsync a file of 10gb between an NT host running Cygwin 4.10 and a Solaris 8 host running rsync 2.5.1 but am having problems with what appears to be a 2gb file limit - the truncation point occurs at 2096111616 bytes. The target filesystem definately supports files over 2GB. Any ideas how to overcome this limitation? Regards, Craig Donnelly Andel Consulting 10 Fenchurch Avenue

What is the default block size? I have a few files 30+mb and data is just added to the end of them. It seems like it takes longer to sync them that it was to send it initially. Should I change the block size or something else? I am running: rsync -z -e ssh *.* user@linuxbox:data/ I need to use ssh because I am going over the internet and sending "company data". -------------- next

Hi List, Any idea how to get wireless lan : 03:00.0 Network controller: Intel Corporation Centrino Advanced-N 6205 (rev 34) working on centos 5? Eero