Re: [Asterisk-Users] No sound on 10% of incoming calls

What do you do with the other 15 channels? your zapata.conf says: channel = 1-15 ;,17-31 = only 15 first channels on PRI but your zaptel.conf says: span=1,1,0,ccs,hdb3 bchan = 1-15, 17-31 You use all 30 channels in Zaptel.conf but only 15 in zapta.conf I never configured Zap on asterisk and

[Asterisk-Users] TEST

Just a Test ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: SV: [Asterisk-Users] callback script?

You can get asterisk source from http://www.asterisk.org/ Arne Morten Johansen wrote: It's a sip channel. http://www.asteriskguru.com/tutorials/unknown_codec_received.html This might work, but I don't know where to find the source-code of asterisk. I've used the ebuilds in gentoo portage to

Re: [Asterisk-Users] BAD/GOOD Echo Cancel

Did you test the echo delay? will 64ms be suffitient? You can easily test the delay by recording the transmit and receive path to a sound file and using some sound editing software see how big the delay is. That is how I did it when I worked for a Telco in Switzerland on theis TDM switch they

Re: [Asterisk-Users] BAD/GOOD Echo Cancel

Doug Lytle wrote: Krystian Filiks wrote: Did you test the echo delay? will 64ms be suffitient? You can easily test the delay by recording the transmit and receive path to a sound file and using some sound editing software see how big the delay is. That is how I did it when I worked

Re: [Asterisk-Users] echo cancel from telco

That is not the whole trough. First you have to make sure that the echo is'nt generated locally by your server hardware, headset, handset, software etc First then you can go to your provider and tell them that you hear echo. The post that you read is talking about the principle of echo

Re: [Asterisk-Users] SIP and NAT - best practices?

Apart of what everyone writes with the NAT=YES I would suggest using canreinvite=no as well as normally asterisk cans the reinvite and this might cause the audio not to get through the NAT and cause dead air for the users specially if the users are behind 2 seperate NAT servers eg. different

RE: [Asterisk-Users] Asterisk Hardware recomendation

. If you would go for plain g711, you could do 500, but i don't recommend it, especially if you have little asterisk experience. (i'd say go for a cluster). Zoa www.asteriskguru.com Krystian Filiks wrote: I will be using IP Hard and soft phones all the way, so everything will be on Ethernet

[Asterisk-Users] Asterisk Hardware recomendation

Hello asterisk people! I have been running a test * server a P III box for some time now and its been rock stable. Now Im looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors. Im wondering what you are thinking about Supermicro 6014H-32

RE: [Asterisk-Users] Asterisk Hardware recomendation

- 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote: Hello asterisk people! I have been running a test * server a P III box for some time now and it's been rock stable. Now I'm looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz

[Asterisk-Users] music on hold trouble

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no

[Asterisk-Users] ASTCC Auth and Dialing problem

Hello, Nobody ever responded to this, does anyone know a solution or has been in a similar situation with ASTCC? Below original email Hi All! Maybe someone can help me.I have 2 problems. I have setup my ASTCC to receive the card number and phone number todial in one string from my sip

[Asterisk-Users] ASTCC one stage dialing problem

Hi All! Maybe someone can help me. I have 2 problems. I have setup my ASTCC to receive the card number and phone number to dial in one string from my sip phone. I dial the destination number and ASTCC authenticates the call using the CALLERID as card number then I want ASTCC to

[Asterisk-Users] SendText application

Hi All How do I use the sendtext app. In asterisk, what is the syntax? I would like to send a text message to a SIP phone when a specific extension is dialed. Thanks KF ___ Asterisk-Users mailing list

[Asterisk-Users] Caller ID on H323

Hi Friends I have a problem presenting Caller ID on my H323 GW. Scenario: Sip Phone Asterisk H323 GW PSTN (E1) From PSTN to the Sip phone works fine I put this lines in extentions.conf exten = 1234,1,SetCallerID, ${CALLERIDNUM} exten = 1234,2,Dial(${testphone1},20,Ttm)

RE: [Asterisk-Users] ASTCC Trunks

Presently, ASTCC does not have builtin support for H323. It would not be hard to add but... If you have a local context that will route the calls for you, you can use the local trunk and send the calls to the appropriate context. Does this make sense. Darren Krystian Filiks wrote: Hi all I

[Asterisk-Users] ASTCC Trunks

Hi all I have asked this question before but have not got any helping input. Im really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand

RE: [Asterisk-Users] Sip Notify and PHP AGI

Hello, Does anyone have any input to my question? Please, this is really urgent. Thanks KF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Krystian Filiks Sent: Monday, January 24, 2005 12:35 AM To: asterisk-users@lists.digium.com

[Asterisk-Users] Sip Notify and PHP AGI

Hello * Users. I need to be able to generate a Sip Notify message using PHP AGI but have no idea how I can do that. What I need to send is the balance of the prepaid card and display it on the soft phones display. Does anyone know how to do this? Thanks in advance. KF

[Asterisk-Users] ASTCC config Problem

Hello all, Im a little stuck in getting astcc working. I went to configure, set up the DB, Created Tables, but what now? What does the Users_Configure, Brands, Cards,Trunks, Routes, Sip Friends etc do? How do I configure Users (In the Users_configure I get Not Configured) How do

[Asterisk-Users] H323 and ASTCC

I got My ASTCC kind of working, but the problem I have is that it tries to send all the calls over SIP. How can I configure it with H323? Thanks KF ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] ASTCC single stage + no access number + auth using sip username and password

Hi All, Im new to * I wonder if anyone have an idea how to make the following with ASTCC. I would like to have all SIP phones to work on prepaid basis and without need to dial any access number, instead I would like to use the phone as normal dialing only the destination number,

[Asterisk-Users] Asterisk ------- Quintum SIP Registration

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

[Asterisk-Users] chan_oh323 loading error

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423

Re: [Asterisk-Users] chan_oh323 loading error

Hi all! Any one that could give me some input on the problem below? regards Krystian Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323

Re: [Asterisk-Users] chan_oh323 loading error

Would the command make P_PTHREADS=1 opt do the job? Krystian Ryan Wilkins wrote: You might try setting P_PTHREADS=1 in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: I have compiled chan_oh323

Re: [Asterisk-Users] chan_oh323 loading error

I have searched through the Makefile and the configure files and could not find any instance of P_PTHREADS. Should I put it there? in that case where? Ryan Wilkins wrote: No. Look in the Makefile of the oh323 driver source. Search down through the file for #P_PTHREADS=1 and remove the #. Then

Re: [Asterisk-Users] chan_oh323 loading error

Ryan Wilkins wrote: Maybe you are not running the latest version of oh323.. I'm running 'asterisk-oh323-0.6.3a'. That is what I'm trying to get going as well with openH323 1.13.5 and pwlib 1.6.6 ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] __use_ast_pthread_create_instead__

Please anyone, When I start * after installing the asterisk-oh323-0.6.3a I get [chan_oh323.so]Aug 15 22:36:44 WARNING[1076252800]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 22:36:44 WARNING[1076252800]:

[Asterisk-Users] H323 problems

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial(SIP/sj1-4ff7, H323/0797617729) in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default,

RE: [Asterisk-Users] H323 call dropped when answered

Like you suggested I tried the g.711 now and got the same, The called number rings but when answered it dropped. I connect to a Quintum Tenor DX. The part I'm curious about is 6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans admissionRequest rejected: requestDenied

RE: [Asterisk-Users] H323 call dropped when answered

Can anyone tell me what this means? 0:54.517 H225 Caller:8141ae8 h323pdu.cxx(1213) H225 Write PDU failed (32): Broken pipe and why this might happen, my call is dropped just after receiving this Thanks /Krystian ___ Asterisk-Users

[Asterisk-Users] H323 call dropped when answered

Hi All. Im using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) - Asterisk --- H323 GK PSTN I have tried all codecs and always the same result, the called phone will ring without dropping for how

[Asterisk-Users] Urgent help with Sip ------ H323 on FREEBSD

I need some help with getting the following to work SipPhone -- Asterisk -- H323 GK (quintum) And H323Phone -- Asterisk -- H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk