Re: [Asterisk-Users] SIP/H.323 HardPhones

2005-11-17 Thread pdhales
Use SIP. PaulH - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, November 18, 2005 7:53 AM Subject: [Asterisk-Users] SIP/H.323 HardPhones > hi, > > Do anyone know a low-cost, simple SI

[Asterisk-Users] SIP/H.323 HardPhones

2005-11-17 Thread [EMAIL PROTECTED]
hi, Do anyone know a low-cost, simple SIP or H.323 hardphone that works well with Asterisk? Jan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/l

Re: [Asterisk-Users] SIP => H.323 Terminator

2005-11-15 Thread Reli Loin
hello, I using asterisk-oh323 i have a problem for send dtmf see log error: reason 24 (Call ended with Q.931 cause [28 - Invalid number format]) thanks for you help 2005/11/15, Martin Vit <[EMAIL PROTECTED]>: > i would recomend this channel for h323: > http://www.inaccessnetworks.com/proje

Re: [Asterisk-Users] SIP => H.323 Terminator

2005-11-15 Thread Martin Vit
i would recomend this channel for h323: http://www.inaccessnetworks.com/projects/asterisk-oh323 Abdul Lateef wrote: Hi all, I have H.323 Terminator and i want to terminate our all SIP clients to this terminator, Is it possible to add H.323 Terminator in Asterisk? Please give me a little hint o

[Asterisk-Users] SIP => H.323 Terminator

2005-11-15 Thread Abdul Lateef
Hi all, I have H.323 Terminator and i want to terminate our all SIP clients to this terminator, Is it possible to add H.323 Terminator in Asterisk? Please give me a little hint os i can start to configure. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +97

[Asterisk-Users] SIP/H.323 suggestion

2005-11-09 Thread Abdul Lateef
HI all, Is Asterisk able to work as SIP and H.323 Gatekeeper same time? If it has the capability to work which i should open? Yours suggestion will be high appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul

Re: [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.

2005-06-13 Thread Damian Minkov
Try playing with faststart . Moises Silva wrote: Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is ma

Re: [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.

2005-06-13 Thread Moises Silva
Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made? best regards On 6/11/05, Carlos Alberto Lara de

[Asterisk-Users] SIP-H.323 dial tone and busy tone problem.

2005-06-11 Thread Carlos Alberto Lara de Hoyos
Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729---

Re: [Asterisk-Users] SIP - H.323 connection

2004-10-26 Thread Luís Palma
> Who is going to accept the SIP calls, Asterisk ou SER? Either one of them can accept/register SIP agents, although what I have been reading SER is more scalable for large implementations. > Who is going to redirect calls to PSTN? Asterisk or SER? Only Asterisk can interface with PSTN using spec

[Asterisk-Users] SIP - H.323 connection

2004-10-21 Thread Joao Pereira
Hello I want to have a server that supports SIP and H.323 endpoints calling eachother, so I installed GNUGK, SER and Asterisk. GnuGK is needed for H.323 endpoints and users Accounting, Authentication and Authorization. SER is needed for the same 3 A's but for SIP users. Asterisk has some goo

Re: [Asterisk-Users] SIP <->h.323

2004-08-14 Thread Bernie Hoeneisen
Hi Ryan! Interesting what experience you have made in this issue. We have setup the alternative channel for H.323 (the * built in chan_h323), and we are now in a testing phase. I was wondering (in case no transcoding is needed), how your setup treats the RTP streams. Do the RTP streams go end-to

Re: [Asterisk-Users] SIP <->h.323

2004-08-13 Thread Ryan Wilkins
Yes, it can.. I'm doing it at my home. My current setup is Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP telephone adapter as my end point talking to Asterisk. For processing sake, you may want to keep

[Asterisk-Users] SIP <->h.323

2004-08-13 Thread Yiannis Costopoulos, Web2Net Solutions Ltd.
Hi, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update optio

[Asterisk-Users] SIP - H.323 Gateway

2003-10-03 Thread Mireia.Munoz-de-jesus
Hi all! Are there some people who have already implemented a SIP - H.323 Gateway? I am trying to do so... but I don't know how. Please if anyone can help me... Thanks a lot for all your answers. Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] SIP/H.323 Phone with intercom

2003-07-24 Thread Peer Oliver schmidt
Good day, anyone had success finding a SIP or H.323 phone with intercom capabilities? Softphone would be preferred? TIA rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users