[asterisk-users] wip5000 crash AP

Good day all I have about 26 Hitachi WIP 5000 They all connect to the 4 Senao Long range AP's 11mb They all have the same ssi but 2 runs on channel 11 and 2 on channel 1 This way the roaming works well! We added a UPS and got POE injectors for each AP BUT..for some reason each now and the

[asterisk-users] wip5000 roaming

Good day all I cant get my WIP 5000 to roam 100% I have 2 access points, different SSIs I make a config1 and config2 on the phone, each for the different SSIDs(A B) Im standing next to A and I walk to B, butthe phone does not want to change its signal to B, it still keeps the bad

RE: [asterisk-users] wip5000 roaming

from there. I would like to hear your results with these phones, is everything working great besides the roaming? On 11/9/06, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I cant get my WIP 5000 to roam 100% I have 2 access points, different SSI's I make a config1 and config2

[asterisk-users] best gui

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time

[Asterisk-Users] how many oh323

Good day. I configured asterisk and oh323.Im using it as a sip-h323 convertor A call will come in to the asterisk box via IAX and be send to a quintum h323 gateway. in oh323 you can set the max in,out and simultaneous calls, Ive set them all to 100. Calls coming in via iax is alaw and then

[Asterisk-Users] cdr server

Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus ___ --Bandwidth and Colocation sponsored by Easynews.com --

Re: [Asterisk-Users] qozap junghanns errors

. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17

Re: [Asterisk-Users] qozap junghanns errors

cards (eth0 for example). We are using Dell PCs but they do not let us to choose how to set interrupts, maybe your PC can. I'm sorry I cannot be more exaustive but this kind of problem is very hard to solve. Giorgio. Altus Snyman wrote: Only one card:-) This is the second time I had

Re: [Asterisk-Users] Voice mailbox on the fly?

Why not exten = 123,1,BackGround(whatIsthe6Digets) exten = 123456,1,Voicemail(u123456) Jim Archer wrote: Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to

Re: [Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap'

I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card

[Asterisk-Users] qozap junghanns errors

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio

Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge

The 1ste pc I tried it on was on a expensive intel board and the second one that worked was on some cheap name board Ill say incompatibility ? Yes, I do use latest bri-stuff package (asterisk 1.0.9 incl) Any ideas? - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526

Re: [Asterisk-Users] chan_capi error2

On Fri, 2005-05-20 at 13:35, Armin Schindler wrote: On Fri, 20 May 2005, Altus Snyman wrote: Good day all I get chan_capi 0.3.5 and I got the patch but when I try make it gives I already asked: What patch do you apply? this error {standard input}: Assembler messages: {standard input

Re: [Asterisk-Users] chan_capi error2

and incomingmsn is what is send to extensions.conf? Thanks again Altus On Fri, 2005-05-20 at 14:32, Armin Schindler wrote: On Fri, 20 May 2005, Altus Snyman wrote: On http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI it tells u if u use the cvs as of april you need a patch I

[Asterisk-Users] chan_capi patch eicon

Good day all Im trying a eicon 4bri card On fedora core 1 I installed the rpm,lsmod says the driver is working I then installed asterisk 1.0.7 I then download chan_capi 0.3.5 But now it says I should patch it for asterisk So I got the patch..fixed it And did a make and it gives a lot of syntax

[Asterisk-Users] eicon fdc3

Good day all Did anyone get the eicon 4 bri working with asterisk and fedora core 3 Please Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] fdc3 no gsm

Good day all I installed Fedora core3 I also installed mpg123 0.59r but asterisk does not want to play anything..on 2 of my server No BAckgroung,Voicemail..nothing Never had this before In the cli it shows its playing it But nothing happens? Please Help

[Asterisk-Users] 2 servers via PRI

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to pri_net...this cant be all? And the cable pin1 -- pin4 pin2 -- pin5

[Asterisk-Users] cdr!

Good day all I installed asterisk-addons and now its logging nicely in my database But I want it to log in my usual log csv as well Please Let me know Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] qozap(!) problem

Same..8a On Mon, 2005-05-09 at 17:12, Eugenio De Vena wrote: Which version of * and bristuff did you install, I had bristuff-0.2.0-RC8a and now I am trying bristuff-0.2.0-RC8c - Original Message - From: Altus Snyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non

Re: [Asterisk-Users] Stun codec

I uses to have this when I enabled stun and did not need it On Tue, 2005-05-10 at 16:55, Ronald Wiplinger wrote: I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the

[Asterisk-Users] asterisk-addon

Good day all I downloaded asterisk-addons to try and make asterisk log in the sql db but when I make a make install i get this error cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:77: macro AST_LIST_REMOVE

[Asterisk-Users] transfer queues agents

Good day all This is what i got off the net about queues and agents Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual

[Asterisk-Users] sangoma fdc 3?

How well does the sangoma cards work with fedora core 3 Im doing the research on what hardware/os I need to use Please help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] qozap(!) problem

Ya well let me know when u solved this We have the same thing Do you have any other cards in with it We have a diguim fxs/fxo card in so maybe its a error with working together Anyway Let me know when you get a fix for it because no one seems to know(or check their /var/log/messages) This lets my

[Asterisk-Users] qozap message error

Good day all with the laster driver and latest drive asterisk I get these errors Please help May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1 71 z2 36 May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 21 z1 30 z2 121 May 3 17:43:15 pbxct

[Asterisk-Users] bri error

Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for

RE: [Asterisk-Users] bri error

signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up

RE: [Asterisk-Users] bri error

signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping

[Asterisk-Users] bri cli error

Good day all I get this error in my cli chan_zap.c:7407 zt_pri_error: PRI: !! Got a UA, but i'm in state 0 I have a 4 port Junghannes card connect with 2 bri isdn lines It keeps on dropping calls and giving errors Please help and advice Thanks ALtus

[Asterisk-Users] security

Good day all I want to put a asterisk server on a public ip and allow any,registered sip and iax connection What security risks are there and how can I secure my pabx One thing I want to know is how do I make it that anyone can call a extension at my box but not make a call out. i.o.w how do I

[Asterisk-Users] analog gsm router

Good day all I have a analog gsm router and a 4 port bri card:-) How do I get the gsm router to work with asterisk I tried adding a voicetronix card but the 2 cards doen not seem to work together,it gives a unresolved symbols error when starting up Any Ideas Please Can you add 2 zaptel

[Asterisk-Users] hangs pc

Good day all I installed asterisk on a pc with redhat 9 and a 4port bri eachtime a call comes in,iax,sip,pstn it just hangs the pc Top shows 75% of the cpu goes to asterisk? Any Idea why? Please Help ___ Asterisk-Users mailing list

[Asterisk-Users] qos test

Good day all I'm looking for a type of QOS test tool(software) I want to test if a link is good enough for voip and test witch ones will be the best..ens any ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] pbx to asterisk

Good day all I just want to know if someone tried this and with out any hassles What I want to do is take 4 extension(analog) of a current,old,pabx unit and put them into a asterisk server with a 4port analog card,like the voicetronix openline4 card. (PSTN)(old PABX)---===(4 ports

[Asterisk-Users] voicetronix bri

Good day all Will a voicetronix openline 4 card work with a 4port BRI card? Please HElp/advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] voicetronix bri

Voicetronix will only be used for the gsm cell router and BRI for outgoing-incoming calls On Thu, 2005-04-14 at 11:26, Michael Bielicki wrote: In what sense ? voicetronix is analog BRI is ISDN digital On 4/14/05, Altus Snyman [EMAIL PROTECTED] wrote: Good day all Will a voicetronix

[Asterisk-Users] voicetronix dtmf

Good day all I got the latest cvs asterisk But when making a call out threw the voicetronix openline4 card the dtmf doens not work I got this in vpb.conf ecsuppthres = 4096 indication = 1 dtmfidd = 3000 ast-dtmf-det=1 relaxdtmf=1 break-for-dtmf=yes Please help Thanks Altus

Re: [Asterisk-Users] fedora 3

Thanks for the trouble n Wed, 2005-04-06 at 15:00, iMRAN wrote: Hi, I`ve installed on FC-3 last month and its working gr8... no probs so far Imran On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have a Fedora core 3 installation Is there any

[Asterisk-Users] Planet VIP 450

Good day all Did someone get the planet VIP 450 working Thanks Altus ___ 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] snom220

Good day all I'm looking for someone with good knowledge of the way the snom220 transfer I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to 200 1st and the transfer B

RE: [Asterisk-Users] snom220

Does Call join on Xfer (2 calls) be on or off? Thanks On Fri, 2005-04-01 at 04:29, Damon Estep wrote: I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to

[Asterisk-Users] sox

Good day all I previously tried the Monitor app with sox but it did not work and according to the list it was because of a broken version What are a good and working version for the latest asterisk Thanks altus ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Fax and Voice

google asterisk fax On Thu, 2005-03-24 at 11:53, Guy Decarpentrie wrote: Hi all, Is * able to do the difference between Fax and voice, and then adapt the treatment of the call ? An example ? Thx ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Fax and Voice

exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit : google asterisk fax Well, i know how

Re: [Asterisk-Users] Fax and Voice

sorry exten = fax,1,Dail On Thu, 2005-03-24 at 12:53, Altus Snyman wrote: exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars

[Asterisk-Users] snom220 problem

Good day all I have a snom 220 with the extra keypad When more than one call comes in none of the extra lines on the phone lights up or anything.You hear the beep in you ear but no way of picking it up.I tied 4 different firmware versions.On was a very old one,with actually worked but is gave echo

[Asterisk-Users] snom 220 version

Good day all What is a good stable snom 220 firmware version. Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Using Codec G-726

had the same thin with 729 I had to go disallow=all allow=g279 On Thu, 2005-03-17 at 16:37, Matt wrote: Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work... ___

[Asterisk-Users] snom 220 busy all the time

Good day all We have a snom 220 that for some reason keeps on giving this message Got SIP response 486 Busy Here back from 192.168.21.222 even though there is no active calls to it and there are 2 accounts set on the phone? Please Help and advice Thanks Altus

[Asterisk-Users] from sip to asterisk to h323..how

Goo day all This is our setup Client phone--(SIP)--asterisk server---SIP/IAX---asterisk--- -- goes out to international server running sip/iax But now I want to dial out to H323 server? I.O.W I want asterisk to act as a H323 client that will rout some calls out to a H323 server.How do I do this

[Asterisk-Users] iax,trunking,zap

Good day all Why do I need a Zaptel card to do trunking in IAX?? What if I only had a voice/iax router? Is there a way around this? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] IAX+G729a

Good day We are going to add 6 channels of G729a to our asterisk server running iax between them I have a few question about the hole license thing. In iax.conf do i allow g729 or g729a?What's the difference? This license is for 2 servers,i.o.w 3 per server.How many calls does this give us? For

[Asterisk-Users] snom220 *8 hangup

Good day all We have a snom 220 set as a switchboard phone I also configured *8 so that if the operator is somewhere else and it rings she can just go *8 on the nearest phone,Grandstrams bt-100 and snom 190.But If she does this she only speaks for about 30s and it will cut off the caller? Any

[Asterisk-Users] hylafax

Good day all Can hylafax work with asterisk..and how I'm trying to find a way to send a fax over my E1 connection Please Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Difference between E1 and PRI

PRI comes in 2versions E1 European and T1 US E1 30 channels T1 23 channels On Wed, 2005-02-23 at 14:15, Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the

Re: [Asterisk-Users] does asterisk support menus?

Yes Application Background() On Tue, 2005-02-22 at 14:35, Muhammad Muzzamil Luqman wrote: Whenever some call comes in i want it to be automatically picked up and then it plays some message Welcome to xyz, Press 1 for sales and 2 for support and then it takes it to the particular extension of

[Asterisk-Users] send fax with pri

HI all What is the best to send a fax with a PRO. I got it working on the receiving and e-mailing it.How do I send one Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] route outgoing call

Good day all I registered at a few sip server in different countries Now I want to route outgoing calls for that country threw that sip server and all the others there my own pstn,ZAP card.I already registered asterisk with them. How would my extensions.conf look.This is what I have but no matter

[Asterisk-Users] Sangoma A101

Good day all Is there any difference in the sangoma zaptel.conf and zapata.conf then other cards Thanks altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

Re: [Asterisk-Users] Sangoma A104 - D-Channel problem

While on sangoma We are getting a samngom pri?Is there any driver I need to install,how does it work,like a Zaptel card. Any doc Please Let me know altus On Fri, 2005-02-18 at 11:06, Kumak wrote: On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote: upgrade to the following

[Asterisk-Users] asterisk qualified

Good day all Is there any time of VOIP/SIP/asterisk qualifications or certificates? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] h323

Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] spandsp asterisk 3/5

Good day all I want to know with version of spandsp works well with ether asterisk 1.0.3 or 1.0.5 Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] asterisk in New-Zealand

Good day all Anyone doing asterisk in New-Zealand? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Asterisk in Singapore.

I can get you a good deal if you import the from South-Africa..Let me know.Altus On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote: In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta

Re: [Asterisk-Users] Bri problem

Thanks Will have a look On Fri, 2005-02-11 at 09:59, Edin Kozo wrote: Hi Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Hope that helps you --- Altus Snyman [EMAIL PROTECTED] escribió: Good day all I've installed

Re: [Asterisk-Users] Cisco7960/SCCP Transfer Help?

If you select more there Trnsfer and BlndXfer will be displayed BlndXfer for Blind transfer Trnsfer for Confirm transfer This is on 7960 On Thu, 2005-02-10 at 15:09, [EMAIL PROTECTED] wrote: I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest

[Asterisk-Users] Bri problem

Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do

[Asterisk-Users] limit iax calls

Good day all We have 2 asterisk servers,connected with iax2 and the phone via SIP They dont have a very big line so I want to restrict the call limet to 3 iax2 calls at a time,and for instance it the 4th call is made it will say something like all lines are being use try later Please help thanks

Re: [Asterisk-Users] How to xfer calls or is my setup wrong?

What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to

[Asterisk-Users] spandsp

Good day all I have a asterisk installation,1.0.3, and spandsp. I got asterisk working,I edited the make file myself. Now when I receive a fax I only get half a page or nothing any Ideas why Please let me know Altus ___ Asterisk-Users mailing list

[Asterisk-Users] bri dropping calls

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

Re: [Asterisk-Users] bri dropping calls

O did not have a look at it yet,I got the one from a week ago,how is aterisk 1.0.5? On Wed, 2005-02-09 at 08:04, Michael Bielicki wrote: hmmm the latest bristuff uses asterisk 1.0.5 so it can't be laast, can it ? cheers Michael On Wed, 09 Feb 2005 07:24:34 +0200, Altus Snyman [EMAIL

[Asterisk-Users] sip_notify.conf

Good day all What is the file sip_notify.conf for Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] bri dropping calls

Where do you get this new version of bristuff,I had a look on the webpage and there's only RC3 On Wed, 2005-02-09 at 08:58, Peer Oliver Schmidt wrote: Altus Snyman wrote: We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3

[Asterisk-Users] warning message

Good day all.I get the warning message on my system,this is for a snom 220,it repeats this message a few times,please help Feb 8 09:29:26 WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Non-critical Request) Is there a page that

Re: [Asterisk-Users] snom soft phone

Did you try 00 That is what it is on the 220 On Tue, 2005-02-08 at 09:36, Paradise Dove wrote: what is the password for Administrator in the softphone? On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke [EMAIL PROTECTED] wrote: Go to the web page, in Preferences there are two

[Asterisk-Users] why asterisk and ser

Good day all Why would u use asterisk and ser together and what is the big difference? Thanks altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] BRI only 2 calls

Good day all I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3 This is to install my quad bri card All installed well I coped over some old config files.All 4 ports are available,so that gives 8 open lines for incoming or outgoing,correct me of I'm rond The problem is,asterisk

[Asterisk-Users] asterisk remote monitor

Good day all We have a few remote pbx systems running I would like to monitor the and check that they are up and running and working Please Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Dialplane slip

Good day all My extensions.conf is something like this [main] ;---incoming+ play welcome message extens = s.. ;---users extensions exten = 100. ;---outgoing ignore 0 ;- It all works fine The message says dial 1 for this ens But if I dial 0+number it will actually make

[Asterisk-Users] h323

Good day all I have a asterisk server running sip and sip phone How do I get asterisk to call another h323 server? Please Help Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] h323 client

Good day all Just to re phrase my previous question We have asterisk running sip for sip phone In the US there is a h323 server What I want to do is: All calls coming into my pbx via sip thats got a american number to go threw the h323 server I have set this up with 2 sip servers where the one

[Asterisk-Users] Grandstreams+Nat

Good day all I cant get my grandstream bt-100 to register My asterisk is on a public ip and the phone behind a nat firewall I added nat=yes in sip.conf and did this on my grandstream set the GS to SIP server=asterisk.yourhost.com and leave Outbound Proxy empty * set the GS to SIP port 5060 and

[Asterisk-Users] ilbc high bandwidth

Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well Is there a way to bring it down? Pleas Help Altus ___ Asterisk-Users mailing list

[Asterisk-Users] sip-sip

Good day all We have a asterisk server running sip for about 20 users We have a client running a unknown sip server in a different country I phone the guy there and he gave a a account(username+password) What I want is if a users calls the number of that country it should be send to the sip server

[Asterisk-Users] Grandstream bt-100 loosing it!

Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]:

[Asterisk-Users] snom220

Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can

Re: [Asterisk-Users] snom220

Sorry It works Just had to reboot the phone On Thu, 2005-01-13 at 08:40, Altus Snyman wrote: Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes

[Asterisk-Users] error?

Good day all I'm getting this error out of the blue on a incoming call? Any idea?Pleas Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ILBC since our native format has changed to SLINR ___ Asterisk-Users mailing list

Re: [Asterisk-Users] fax e-mail spandsp

Did anyone get asterisk to actually work with a fax coming in on a pri number and e-mail it to a user? On Mon, 2005-01-10 at 08:29, Howard Lowndes wrote: On Mon, 2005-01-10 at 16:00, Altus Snyman wrote: Its still fails! [EMAIL PROTECTED] apps]# patch apps_makefile.patch.new patching

Re: [Asterisk-Users] fax e-mail spandsp

the changes in the apps/Makefile have progressed while the patch makefile have not. Here is a current patch that works as of CVS-HEAD-01/06/05-14:47:06 Regards, Jim On Fri, 7 Jan 2005, Altus Snyman wrote: I'm trying to install spandsp But when I try to patch the Makefile it gives

[Asterisk-Users] TE110P error

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10

RE: [Asterisk-Users] TE110P error

? A better guess is either the driver for the card isn't loaded or the zap config files aren't agreeing with each other. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: Monday, January 10, 2005 1:24 AM To: asterisk Subject

[Asterisk-Users] fax e-mail spandsp

I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is

[Asterisk-Users] fax to email

Good day all I have a pri card,e100 What I want to do is If a fax comes in for number 1234567890 it should be e-mail to [EMAIL PROTECTED] If a fax comes in for number 0987654321 it should be e-mail to [EMAIL PROTECTED] ens Can this be done and how

Re: [Asterisk-Users] fax to email

and email-fax?? The other way around On Thu, 2005-01-06 at 14:17, Andrew Kohlsmith wrote: On January 6, 2005 06:55 am, Altus Snyman wrote: Good day all I have a pri card,e100 What I want to do is If a fax comes in for number 1234567890 it should be e-mail to [EMAIL PROTECTED

Re: [Asterisk-Users] fax to email

How do I fax a .tiff file with asterisk? On Thu, 2005-01-06 at 15:13, Michael Welter wrote: Altus Snyman wrote: and email-fax?? The other way around You can run a simple mail server on the * box to accept emails addressed to the .fax domain (i.e. [EMAIL PROTECTED]). This presumes

[Asterisk-Users] Call(out) routing

Good day all I had a look at the extensions.conf sorting http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting What I'm trying to do is route all my cellphone number threw a channel and all other calls threw the other 3 channels Cellphone numbers are 10 number,i.o.w XX.

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