search for: ht496

...65fffff From: <sip:435572949012@194.208.44.43;user=phone>;tag=f8e70000a6870000 To: <sip:435572949012@194.208.44.43;user=phone> Contact: <sip:PPC998202@194.208.44.44:34560;user=phone> Call-ID: e62dffffcd4dffff@194.183.145.211 CSeq: 100 REGISTER Expires: 3600 User-Agent: Grandstream HT496 1.0.2.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 0: REGISTER sip:194.208.44.43 SIP/2.0 (34) Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 1: Via: S...

I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards,

Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of those two extensions if: I defined both as ulaw (G.711) One as ulaw and the other as G.729 Only one at the time if I define both as G729 Only the G711 if I define one as G726 and the other as G711. No way to make calls...

At 09:37 AM 12/25/2006, you wrote: >The Asterisk Development Team is pleased to announce the first >release in the Asterisk 1.4 series, Asterisk 1.4.0! Another thing I've noticed is that twice today while sitting at the CLI prompt I was throw to the command line because Asterisk had exited. Typing asterisk started it right up again and then I could get back to the CLI. I think both

...hope this help someone else... Today before to find the new information: Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of those two extensions if: I defined both as ulaw (G.711) One as ulaw and the other as G.729 Only one at the time if I define both as G729 Only the G711 if I define one as G726 and the other as G711. No way to make calls...

Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35 DEBUG[27914] chan_sip.c...