Re: [Asterisk-Users] New Firefly version

adam - can the g729.dll be downloaded somewhere - is this still required for g.729 support? Regards, Steven Thomas jo [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 31/05/2004 09:19 PM Please respond to asterisk-users To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users

[Asterisk-Users] Pulver WiSIP Dual Line and Hold?

Hi, I have received my WiSIP phone - works well for basic functions of call answer and hang-up! Does anyone know how to enable Dual line support, Hold and Transfer functions with this phone via Asterisk. Thanks, Regards, Steven Thomas

[Asterisk-Users] BCM Wireless SIP Phone

Hi, Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm Regards, Steven Thomas Network Integration Services IBM Australia Ph: 0404 099 262 NH011, IBM Centre, 601 Pacific Hwy, St Leonards, 2065.

Re: [Asterisk-Users] Cisco Gateway Integration

yes. Cisco 2612 Router with 2 x FXO's and 2 x FXS's. Works well using H323, and gnugk. Steve. Bruce Hedreen [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 15/12/2003 09:57 AM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: [Asterisk-Users]

[Asterisk-Users] RTP Codec Error(s) - Is there really a solution for this or these?

, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its

[Asterisk-Users] Call logging In and Out

Is it possible to log the CallerID of an inbound call including the time to a log / text file? Also the same for outbound? ie., dialed number and time? Thanks. Regards, Steven Thomas

Re: [Asterisk-Users] delay problem in h323

I assume it manages the signal part of the RTP stream but not the RTP voice stream at the codec level? Maybe someone else can comment on the translation methodologies within Asterisk? Regards, Steven Thomas

Re: [Asterisk-Users] delay problem in h323

- not sure why! Regards, Steven Thomas andrea [EMAIL PROTECTED

Re: [Asterisk-Users] delay problem in h323

also. I would suggest trying chan_h323 as an alternative. Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its

[Asterisk-Users] CallerID through the GnuGK - does this work?

Hi - can anyone confirm or deny that CallerID works through (passes through) the GnuGK? ie., X100P - Asterisk - GnuGK - Gateway Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] Chan_h323 support for phone numbers via gateway?

Does chan_h323 support phone number calling via a gateway? ie., something like calling 5000 forwarded to: exten = 5000,1,Dial(h323/[EMAIL PROTECTED]) if so - what format should the exten be in? Thanks. Regards, Steven Thomas ___ Asterisk

[Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway

fine if it is just an IP address that it is calling, ie, a softphone. Thanks for your help Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323

I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas

Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323

Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas

[Asterisk-Users] Chan_h323 one way audio

, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] No voice call from H.323-phone to SIP-phone

Hi, Did anyone have any comments on the below problem - or did you (shong ching) manage to solve this? I have the same issue - any assistance would be great. Thanks. Regards, Steven Thomas

Re: [Asterisk-Users] Chan_h323 one way audio

not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua

[Asterisk-Users] Chan_h323.so native?

Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? Thanks. Regards, Steven Thomas

[Asterisk-Users] Chan_oh323 Dial format / voice latency 4 to 5 secs

Hi, Can someone confirm the format of the Dial string for a H.323 gateway using chan_oh323? The format I have working is: exten = 5000,1,Dial(OH323/h323:[EMAIL PROTECTED]) I have 5000 as a speed dial - the extension functions, but the voice latency within the call to the analog phone

Re: [Asterisk-Users] h323 gateway call lost after 74sec always

Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router

Re: [Asterisk-Users] h323 gateway call lost after 74sec always

Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly

[Asterisk-Users] h323 gateway call lost after 74sec always

Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered