similar to: Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd)

Displaying 20 results from an estimated 2000 matches similar to: "Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd)"

2005 Jun 09
0
Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf?
Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to receive the call: [from-pstn] exten =>
2005 Aug 29
1
grandstream handytone 488 fxo
can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that asterisk@home has this built in to just a button hit, but i dont want to
2007 Mar 07
1
Problem HandyTone 488 does not call transfer
Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833
2006 Nov 02
2
Grandstream HandyTone-488 with Asterisk ?
Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye
2006 Mar 30
2
Connecting a Grandstream Handytone 486 to Asterisk
Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call
2004 Mar 06
2
GS HandyTone-286 Transfer Problem, can anyone confirm?
There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA
2005 May 31
2
handytone 486
Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli
2003 Nov 25
3
Handytone 286 - calling out
Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
2004 Aug 27
1
Help with a fax via Grandstream Handytone 286?
I have an analog Fax machine which I wish to connect to the network and the Asterisk server. It will connect through a GS Handytone 286 converter and then into the LAN. Is there any information out there on what I need to write in *sip.conf* and/or *extensions.conf* to make sure the fax works as a fax? Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do I need to
2004 Jan 16
0
GS Handytone Echo-problem
Hi, Yesterday I finaly got my handytone sip adaptor. It works.... But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to
2004 Dec 01
0
Grandstream BT100 / HandyTone 286 and Level 3
Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes
2005 Aug 11
0
* behind NAT, client behind NAT(handytone 286), very strange behavior
Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext
2006 Jan 29
1
HandyTone 488 ata?
Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?
2010 Apr 29
4
ATA shootout: PAP2T versus Grandstream Handytone 286
I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an
2004 Jun 05
1
FXO answering quicker
Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found a configuration variable to let me do this... Thanks in advance, Andrew _________________________ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568
2007 Dec 04
2
pstn call waiting and zap
Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the
2003 Nov 12
0
Sipura / Handytone / Cisco
Could anybody shed some light in which device they would use in this situation: Remote office PBX's to be connected via a) Cisco ATA-186 or b) Sipura SPA-2000 or c) Grandstream HT-ATA-286 to go via the net to an * box. Pros / Cons for each device would be appreciated! Thanks Kris
2006 May 08
3
PSTN Incoming call on real line disrupts VoIP call over DSL circuit
I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone
2011 Mar 07
0
[fdo] Global architecture ?
Hello, I'm working on an Openmoko FreeRunner where I'm supposed to use the GPS module. I just have a question about the global architecture concerning the different layers from the hardware to the final GPS application. I'm using ogpsd, fso-gpsd, etc. What is "Gypsy" then ? Is this scheme correct : hardware <-> ogpsd (= gypsy) <-> fso-gpsd <-> tangoGPS