Re: [Asterisk-Users] Which PRI card for EuroISDN ?

- Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 19, 2005 4:36 AM Subject: Re: [Asterisk-Users] Which PRI card for EuroISDN ? On Fri, 18 Feb 2005, Robert

[Asterisk-Users] Terminating IAX calls in E1 PRI interface - what do I need to be able to send arbitrary caller id to called party ?

Hi, I'd like to terminate IAX call on PRI interface. What conditions should be met to be able to send arbitrary caller numbers to called party, so he can call back to supplied ISDN number (that is different for every IAX user) - not through PRI interface, but plain ISDN call !! Thanks in

[Asterisk-Users] Looking for Asterisk setup and maintainance (terminating calls to EuroISDN PRI interface) in Frankfurt Germany

Hi, we have a client who seeks for help setting up and maintaining Asterisk server (plain IAX trunk or SIP terminating calls on PRI card - nothing else) in location Frankfurt, Germany. Should be operational in 30 days... Please contact me offlist with offer ... thanks in advance, regards,

[Asterisk-Users] Which PRI card for EuroISDN ?

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Which PRI card for EuroISDN ?

- Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 11:49 PM Subject: Re: [Asterisk-Users] Which PRI card for EuroISDN ? On 23:11, Fri 18 Feb 05, Robert Rozman wrote: Hi, I wonder which PRI interface

Re: [Asterisk-Users] capiECT problem

, Robert Rozman wrote: Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound

Re: [Asterisk-Users] festival text for weather report

Hi, I could recomend GEO::Weather or some similar CPAN module that already connects to proper source and parses info to well organized data structures for weather info . Same could be applied to other sources (News, ...). Regards, Rob. - Original Message - From: Ernie Ankele

[Asterisk-Users] When callerid changes its value ?

Hi, I'm reading a lot of stuff about callerid problems, but couldn't find any logical explanation of Asterisk behaviour with callerid. When I receive incoming call, caller info seems ok, but when transferred to local extension via some macros, callerid gets to 'asterisk'. Does anyone know why and

[Asterisk-Users] capiECT problem

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten =

Re: [Asterisk-Users] speech recognition V 2.0

- Original Message - From: Race Vanderdecken [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 6:57 PM Subject: RE: [Asterisk-Users] speech recognition V 2.0 Greetings David, PerlBox

[Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to

[Asterisk-Users] 4xHFC-s cards vs 1 quadbri HFC-4S card ?

Hi, I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120 EUR) and using 1 QuadBRI card (approx. 700 EUR) ? What makes such difference ? Is it possible to do first configuration ? With what drivers ? Is it stable ? Thanks in advance, regards, Rob.

Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff

- Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:57 AM Subject: Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff On 14:10, Fri 11 Feb 05, Remco Barende wrote: Hi list! I'm currently

[Asterisk-Users] What quad/octo BRI cards are best/stable for EuroISDN and Asterisk ?

Hi, I'm currently deciding on what card to pruchase for octo/quad BRI card to use with Asterisk on EuroISDN lines. I'm aware of at least two options (Junghanns or Beronet), but don't know how stable and well supported they are. Which ones are better supported ? Any experiences? Any advice ? How

[Asterisk-Users] How stable are cheap HFC-s cards in NT mode ?

Hi, I'd like to use one card to interface with existing ISDN pbx output. How stable are those cards for this ? Where can I find more info how to setup ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Proper handling of incoming IAX/SIP callerids to be able to call back - why is calleridnum stripping dots out of number ?

: ${CALLERIDNUM}) ;exten = s,3,SetCallerID(${CALLERIDNUM}) exten = s,5,Goto(from-pstn,s,1) and when executed : -- Accepting unauthenticated call from 193.77.90.224, requested format = 2, actual format = 2 -- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, IAX call from outside Robert Rozman [EMAIL

[Asterisk-Users] How can agent logout manually ?

Hi, I don't know how to logout agent. The trick from Wiki (stated below) doesn't work (I have CVS stable from yesterday). I get invalid login if don't specifiy Agent ID. regards, Rob. Logging off the queue manually 1.. call the extension for AgentCallbackLogin 2.. enter your password

[Asterisk-Users] Can agents login be permanent across Asterisk restarts ?

Hi, I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is restarted. Can this be avoided in some way ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] how to pop up called number details using phpscripts in agi scripts

Hi, Covide looks interesting. Is this a killer combination of groupware and Asterisk I was looking for ? Is it open source ? Do you have any more english info ? Thanks in advance, regards, Rob. - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To:

Re: [Asterisk-Users] Zombie SIP channels

- Original Message - From: Pedro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:08 AM Subject: Re: [Asterisk-Users] Zombie SIP channels Thanks for the tip. They both seemed to go away

Re: [Asterisk-Users] SIPP load testing - unexpected message - anyoneusing sipp sucessfully ?

Hi, I used this command line : sipp -sn uac URL_of_*_server -trace_err Is there any better ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] SIPP load testing - unexpected message - anyone using sipp sucessfully ?

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get Unexpected message for Call-ID ..., so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test

[Asterisk-Users] Asterisk cmd SayNumber : how to pronounce in another language - we say one-and-twenty instead of twenty-one

Hi, I wonder how SayNumber can handle international numbers (I can translate numbers - but would also need different order...). I guess that solution for German language will also work in our native language. Thanks, Regards, Rob. ___

[Asterisk-Users] Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes

[Asterisk-Users] Actions taken drugin calls - are there any other keys active beside # for transfer ?

Hi, I'd like to trigger call recording during call. Do I have any keys that can be pressed during call ? I've tried this, but doesn't start anything ( I guess that a is active only during voicemail ?): exten = a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ; Already recording ? if

[Asterisk-Users] How to mark calls for inclusion in CDR ?

Hi, I'd like to mark calls to get into CDR when they are received (so I'll will be able to mark only incoming or outgoing calls without locals). I thought that maybe doing so would work, but would kindly ask for your opinion or maybe better advice: a.. ${UNIQUEID}: Current call unique

[Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stable asterisk

Hi, I have downloaded files and also local versions of pwlib oh323 (both Janus patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention which local libraries should be downloaded from inaccess to get everything

Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stableasterisk

- Original Message - From: Roger Schreiter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 01, 2005 1:01 PM Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stableasterisk Robert

Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stableasterisk

- Original Message - From: Michael Manousos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 01, 2005 1:23 PM Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stableasterisk Robert

Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvsstableasterisk

- Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 01, 2005 3:29 PM Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvsstableasterisk

[Asterisk-Users] mysql based adressbook with agi and web interface ?

Hi, I'm looking for adressbook that could easily integrate into Asterisk, so it should: - have agi script to search for caller id name - have fields for notes on previous contacts (for CRM spawning of FOP) - have web interface to edit entries easily ... Any advice, pointers ? What is your

[Asterisk-Users] Can I start recording during call - is priority a active only in voicemail ?

Hi, I'd like to trigger call recording during call. Do I have any keys that can be pressed during call ? I've tried this, but doesn't start anything ( I guess that a is active only during voicemail ?): exten = a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ; Already recording ? if

[Asterisk-Users] Meetme2 web - nothing happens on click ?

Hi, I've installed meetme2 according to instructions. Everything seems ok, members of conference are displayed, but nothing happens if I click on 3 action buttons (kick out, talklisten, ...). Any hints how to deal with this ? In what way exactly does meetme2 kick user off the conference ?

Re: [Asterisk-Users] Speech Recognition

Hi, probably I won't be much of help, but I'm also looking for speech recognition solution. But we're actually looking at two problems: - one would be so called voice dialing (similar to celular phones) - one records its own spoken names and speaks them after to call certain person - this problem

Re: [Asterisk-Users] Integrating with existing 1BRI, 6 POTS Panasonic PBX ?

- Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 28, 2005 10:22 PM Subject: [Asterisk-Users] Integrating with existing 1BRI,6 POTS Panasonic PBX ? Hi

[Asterisk-Users] Moh in meetme doesn't work if I transfer to meetme

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc:

[Asterisk-Users] How to check sip channel with smoething similar to ping ?

Hi, I saw Nagios plugin that can check if Asterisk IAX2 channels is alive. Can I do the same with SIP channel ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer tomeetme

Hi, - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 27, 2005 9:10 PM Subject: Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer tomeetme It's hard to tell

[Asterisk-Users] Problems making SIP URL outgoing dial

Hi, I'd like to call my friends through their SIP URLs. I've found two approaches for doing this in Asterisk: - one is to prepend some numbers at start and catch them - the rest of called string is used for SIP URL - another approach (that I like better) is to use catchall pattern at the end of

Re: [Asterisk-Users] Asterisk with Grandstream ringback

- Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I updated to firmware version

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

Hi, first big thanks for all your apps, Areski. I'm using them and I'm really satisfied. But I do use them with mysql and would like to do it with this one too I'm just kindly asking for addition of mysql support... Regards, Rob. - Original Message - From: Robert Augustyn [EMAIL

[Asterisk-Users] Asterisk as root in realtime vs. non-root asterisk ?

Hi, what would be general choice between those two options? They are related to two different things: security, performance. But what would I loose if I run as non-root without realtime priority ? Thanks in advance, regards, Rob. ___

Re: [Asterisk-Users] Asterisk as root in realtime vs. non-rootasterisk?

servers all in non-root and have no problems. -Matthew - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 2:48 PM Subject: [Asterisk-Users] Asterisk

[Asterisk-Users] Void callerid info on iax clients, but OK from local extensions or on SIP clients

Hi, first I wanted BT100 to display caller id properly. But on this way (BT100 now works) I lost caller id on all iax clients (Firefly, iaxphone) - when call comes from outside CAPI line, although according to console callerid is there (only number). Are there any special hints on displaying

[Asterisk-Users] Firefly reject problem - it just keeps ringing

Hi, when I press reject on Firefly, another end hears congestion, but I'm still listening to ringing... Also how you determine which version of Firefly you have? I also don't understand licence completely and would like to get this answer - can I distribute Firefly to commercial users if I

[Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of

Re: [Asterisk-Users] Asterisk with Grandstream ringback

- Original Message - From: Kim Lux [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 7:54 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I've got the

[Asterisk-Users] Asterisk auto-dial out with .call files: Can I provide caller ID to second extension ?

Hi, I'm setting up system with repeated calling of outside extension. When it answers, local extension will ring. Supplied caller id displays correctly on outside phone, but on local extension it's empty. Can I somehow supply proper caller id to local extension too ? Regards, Rob.

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

- Original Message - From: Steve Kann [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 3:56 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone Robert Rozman wrote

Re: [Asterisk-Users] AMP with SUSE 9.2

No, except one minor thing - htdocs home directory is in different place... Regards, Rob. - Original Message - From: Keith Burns [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 4:19 PM Subject: [Asterisk-Users] AMP with SUSE 9.2 Hi, I have

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

- Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone): I suppose you're

[Asterisk-Users] What softphones for commercial use ?

Hi, I wonder what softphones (cheap or free) are best for in production use ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] chan_iax2.c:5441 socket_read: Rejected connect attempt from

Hi, I'd like to get URI dialing from iax clients working. If I call Ip directly (192.168.0.50) it goes straight to from-iax context. But if I call [EMAIL PROTECTED] (200 is local extension) then I get upper message. Is there any setting that is preventing me for calling that extension (I

[Asterisk-Users] Iaxphone - unreachable if qualify yes ?

Hi, if I change Iaxphone settings to qualify=yes it says it's unreachable. Can iax2 clients be monitored with qualify option ? Is this problem related to iaxphone ? Anyone sucessfully using iax qualify feature ? Regards, Rob. ___ Asterisk-Users

[Asterisk-Users] Can I get info about email addresses from voicemail.conf in dialplan or variables ?

Hi, I'd like to setup automatic recording of channels and send wav files via email to extension user (to same email address as in voicemail.conf). Can I access those addresses from dialplan or AGI ? Regards, Rob. ___ Asterisk-Users mailing list

[Asterisk-Users] Can I start recording channel in the middle of conversation ?

Hi, I'd kindly ask for simple example if this is possible ? Is any key press encountered during conversation and action taken in dialplan ? Thanks, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Can I start recording channel in the middle of conversation ?

Hi, I'd kindly ask for simple example if this is possible ? Is any key press encountered during conversation and action taken in dialplan ? Thanks, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] announcing caller id?

Hi, - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 16, 2005 10:20 AM Subject: Re: [Asterisk-Users] announcing caller id? Any one have any solution for this?

[Asterisk-Users] Terminating VOIP calls on EuroISDN PRI interface ?

Hi, I'm looking for HW to terminate VOIP calls (SIP, if IAX2 even better) from remote Asterisks on PRI EURO ISDN interface. One option is PC with Asterisk and PRI card, but are there any embedded devices to do that ? Will they be at same price or more expensive ? Regards, Rob.

[Asterisk-Users] Can I use spandsp with Asterisk on Fritz with Capi ?

Hi, answer is probably obvious, but couldn't find it... Thanks, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] BT keeps open sip channels

Hi, I've switched to fresh install and unfortunately changed two things at the time: - used fresh AMP install - upgraded Grandstream bt100 to latest beta from Grandstream web site .18 I have one local IAX2 extension (IAX phone) and one bt 100 extension. I cannot make calls between those two.

[Asterisk-Users] Problems calling between two local SIP extensions

Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context:

[Asterisk-Users] GSM adapter for analog telephone - connect with fxo or fxs to Asterisk

Hi, I have Siemens combiset - it can gateway GSM phone to normal analog phone. It has output where I can connect regulat analog phone. How can I connect to combiset with Asterisk - via fxo or fxs ? Thanks, regards, Robert. ___ Asterisk-Users

[Asterisk-Users] Any experience with Linksys WRT54GP2 as local extensions to Asterisk ?

Hi, I'd just like to confirm compatibility of Linksys router WRT54GP2 as local extensions to Asterisk. Can it register to local Asterisk behing him ? How stable/good is analog interface ? Any experience would be more than welcome. Thanks in advance, regards, Rob.

[Asterisk-Users] Sip protocol question ...

Hi, I'm tryinig to debug SIP call from activex control based on MS RTC (A) to Asterisk (B). I use Etherreal to follow packages and I would like to ask short questions: - Session trace shows following order of packets: A - BInvite B - A100 Trying B - A200

Re: [Asterisk-Users] Sip protocol question ...

://www.pernau.at/kd/voip/ActXPhone/ Regards, Rob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: vendredi 7 janvier 2005 11:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sip protocol

[Asterisk-Users] Sipura 2000 vs 2100

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ?What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

[Asterisk-Users] Does spandsp work with capi channels ?

Hi, if not, what are my options (beside dedicating one line to capi4hylafax). I'd like to cover faxes and voice on same channells - is this impossible with capi ? Thanks, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Announcements via IAX phones

Hi, I wonder if you're willing to share your setup with autoanswer mode... Regards, Rob. - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, January 02, 2005 7:50 AM Subject: [Asterisk-Users] Announcements via IAX phones

[Asterisk-Users] Can I receive faxes with Fritz card Asterisk ?

Hi, I'm reading that spandsp works only with zaptel channels. What are my options if I want to receive faxes through ISDN Fritz card with Asterisk and possibly forward it as emails ? Regards, Rob. ___ Asterisk-Users mailing list

Re: [Asterisk-Users] How to connect two Asterisks as secure aspossible without too much additional bandwidth ?

much additional bandwidth ? Robert Rozman wrote: Hi, I plan to connect to remote Asterisk that will terminate calls to ISDN primary channel. I'd certainly like to secure this type of service, so would kindly ask for any advice on how to secure this authentication as much

[Asterisk-Users] How to connect two Asterisks as secure as possible without too much additional bandwidth ?

Hi, I plan to connect to remote Asterisk that will terminate calls to ISDN primary channel. I'd certainly like to secure this type of service, so would kindly ask for any advice on how to secure this authentication as much as reasonably possible. Since there is long IP route I guess VPN will

[Asterisk-Users] Which Primary ISDN card to use in Europe ?

Hi, I'd kindly ask for any advices, experiences and opinions on what Primary (and also several BRI) cards should be used on EuroISDN and Asterisk ? I'm also curious what features do these cards offer regarding echo cancellation (mostly when calling from analog lines) ? Thanks in advance,

[Asterisk-Users] Can I read more than 7 numbers from capi ?

Hi, we have 7 numbers in out country phone system. If I dial 123456766 and 1234567 is my number - is there any way to get those two extra numbers in Asterisk for transfer to local extension ? I have AVM Fritz with CAPI driver. Thanks in advance, regards, Rob.

Re: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT -cancall but cannot receive calls ?

breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a phone that can call then a phone that can autoanswer... -Original Message- From: Robert Rozman [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 14

[Asterisk-Users] detected NAT type is full cone for BT behind nat ?

Hi, I wonder what does this warning 399 mean and how to workaround? sip show peers says that sip client is unreachable althought it works with some eexceptions ... I saw posts in this list about setting codec to ilbc, is this right action ? Also, I'm very interested if anyone succeded on

[Asterisk-Users] Asterisk to sip client behind Firewall/NAT - can call but cannot receive calls ?

Hi, I have following setup: BT100 Firewall/nat 1 (www.ipcop.org) Internet Firewall/nat2 (Vigor) Asterisk . I'd like to use BT100 as local extension to Asterisk. I've done simple setup and BT100 can call Asterisk and place outgoing calls. However I cannot set him to qualify,

[Asterisk-Users] How to setup private enum server ?

Hi, I'd like to setup little private enum server. Any more info on how to do that ? Regards, Rob. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] 20 BT-100 setup - what firmware is recomended ?

Hi, I'd like to setup 20 BT-100 with Asterisk. If I got all discussion on grandstreams right, I should put my own tftp server and point phones to it. On phones is 1.0.5.16 firmware. Is this one good or should I up(down) grade to certain version ? What functionality is possible with BT-100 ?

[Asterisk-Users] Many similar contexts - can I use Macro or some other template concept ?

Hi, I'd like to make small 20 users setup with BTs. I'd like each of them to have its own context (for recording prompts, conference, ...). For them to have same extensions I should put them in separate contexts and let BT call them offhook. But these contexts are pretty similar (for instance

Re: [Asterisk-Users] No ring signal when calling internal extensions ?

- Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, December 10, 2004 1:49 AM Subject: Re: [Asterisk-Users] No ring signal when calling internal extensions ? Robert Rozman

[Asterisk-Users] Aditional local number when calling from ISDN thtough Capi to local extension ?

Hi, I have BRI ISDN access. Now I'd like to call from ISDN phone and type phone number of Asterisk server like 123456 and then imediately add numbers for local extensions (like 12345611), so call would go directly local extension 11. I'd like to achieve this functionality through Fritz and Capi.

[Asterisk-Users] Intercept and redirect outgoing calls ?

Hi, I'd like to do simple LCR - when user dials number, would like to check against database and if that number is available over VOIP, simply substitute dialed number with SIP address for instance and call over VOIP. What are possibilities to do that with Asterisk (AGI, manager API) ? And would

Re: [Asterisk-Users] Intercept and redirect outgoing calls ?

- Original Message - From: Nick Barnes [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, December 10, 2004 5:11 PM Subject: RE: [Asterisk-Users] Intercept and redirect outgoing calls ? I'd like to do simple LCR - when

Re: [Asterisk-Users] Intercept and redirect outgoing calls ?

PM +0100 Robert Rozman [EMAIL PROTECTED] wrote: - Original Message - From: Nick Barnes [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, December 10, 2004 5:11 PM Subject: RE: [Asterisk-Users] Intercept and redirect

[Asterisk-Users] No ring signal when calling internal extensions ?

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten = 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't

Re: [Asterisk-Users] No ring signal when calling internal extensions ?

signal when calling internal extensions ? On Thu, 9 Dec 2004, Robert Rozman wrote: Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten

Re: [Asterisk-Users] No ring signal when calling internal extensions ?

PROTECTED] Sent: Friday, December 10, 2004 12:41 AM Subject: Re: [Asterisk-Users] No ring signal when calling internal extensions ? Robert Rozman wrote: Sorry, wasn't specific enough. Caller is not hearing any ringing tone. That means just plain silence til local extension picks up

Re: [Asterisk-Users] G.729 algorithm?

Hi, do you have info in what countries g.729 is not valid... ? Regards, Robert. - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]; Asterisk Developer Mailing List [EMAIL PROTECTED] Sent:

Re: [Asterisk-Users] more than 3 msns with chan_capi

Hi, sorry for newbie Fritz question. I always thought that AVM Fritz has 2 devices for 2 MSNs. So does this mean, that Fritz can handle more ISDN lines ? Does this mean you can have more than 2 calls at once ? What is MAX number of parallel calls ? Thanks in advance, Regards, Robert. -

Re: [Asterisk-Users] Enhanced Audio Support for EAGIs

Hi, I'd just like to say that I'm interested in this thing. Do you intend to use Sphinx 4 ? Can sphinx use HTK HMM models files ? Please keep us posted on progress Regards, Robert. - Original Message - From: Jeff Maki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November

[Asterisk-Users] Running Asterisk in chroot environment ?

Hi, I'd like to get some opinions whether is appropriate to run Asterisk in chroot environment... I'd also like to hear if anyone is willing to share knowledge or scripts to make chroot environment for Asterisk ? Thanks in advance, Robert. ___

[Asterisk-Users] How to connect Siemens Combiset to Asterisk - fxo or fxs ?

Hi, I have Siemens combiset - it can gateway GSM phone to normal analog phone. It has output where I can connect regulat analog phone. How can I connect to combiset with Asterisk - via fxo or fxs ? Thanks, regards, Robert. ___ Asterisk-Users

Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?

Any more info how to configure Asterisk to limit the number of calls concurrently ? Thanks in advance, Robert. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:50 AM Subject: RE: [Asterisk-Users] Can bad person with SIPp attack

Re: [Asterisk-Users] Echo in CAPI channels

Hi, I have exactly same problem. I hear echo (I guess it's more like reverb) but only if calling from analog phone. I've already asked this group about it, but didn't receive any solution answer. Maybe we should try to contact avm or something. I also didn't find any more info what these all

Re: [Asterisk-Users] Echo in CAPI channels

Hi, who does this echo canceling if you use ordinary ISDN phone (you hear no echo) ? Regards, robert. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 1:12 PM Subject: Re:

Re: [Asterisk-Users] Echo in CAPI channels

Please keep us posted on progress. There are more users watching you :-). Regards, Robert. - Original Message - From: Derek Conniffe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 5:08 PM Subject: Re:

Re: [Asterisk-Users] WRT54GS zaptel timing device

Hi, I'm desperately looking for more info about running Asterisk on WRT54GS. Can you please give some more info how to do this (any pointer to site, more info ...) ? How much room is there on router for software ? Thanks in advance, regards, Robert. - Original Message - From:

[Asterisk-Users] Can bad person with SIPp attack Asterisk ?

Hi, sorry maybe dumb question. But could person with bad intent attack Asterisk PBX with SIPp tool ? Can Asterisk be overloaded this way and not working OK for the rest of conversations ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?

specific to Asterisk. Otherwise, they'd have to rely on a security hole in the software itself. I don't know of any, and I'm sure they'd get fixed really fast if they were found... -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman

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