similar to: grandstream bt-100 callerID not appear

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hello, I am moving topic from asterisk-dev list to asterisk-users list. Did anyone succeed receive incoming calls in h323 and orient them to right context based on "host" identification? To summarise, I have quintum Gateway sending call to Asterisk box, and I would like to use asterisk as a protocol converter h323 --> sip. in h323.conf, I have [quintum_gw1] type=user

Hello, Thank you for the job well-done. I installed the chan_h323 of the asterisk-1.2.7.1 and with lib pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed g729 from digium. However, I am having a very funny behavour. 1. If I send a call on its ringing at the called side but the caller didn't get the ringing tone. 2. if the called picks up the phone, I am

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

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

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi list, I have some nontrivial questions. I am no telecommunication guru and I will explain it with my simple words. I hope someone can help me with these issues (with Asterisk 1.0.3): - If I call outside (with Nikotel to German Telekom) there is a remote hangup after 2 minutes. I've seen other people posting this but nothing helped. I luckily managed to get around this issue with the

Hello, I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Regards, Christopher

Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from '<sip:144@192.168.0.250>' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from

>I was successfully using the BT-100 phone with CVS 11/10. Now that I've >upgraded to 11/27, I can't place an outbound call. However the phone is >registered and works well with inbound calls. Any suggestions will be >appreciated. Thank you. Hi! I encounter similar problems. But in my case also incomming calls are not possible. But this might be because of my upgrade to

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

Hi there, for those that haven't yet found out for themselves: - BudgeTone/ HandyTone firmware now has an option for "disable callwaiting" which probably eliminates the most urgent need for outgoinglimit= and incominglimit= in sip.conf (firmware 1.0.4.54 and later, maybe even available in some slightly earlier versions) - new option "subscribe for MWMI" (message

Hey all. I've got a weird problem with the grandstream budgetone101 and asterisk that I'm having no luck finding any info on. I'm positive it's a grandstream problem but i'm hoping someone here can at least point me in the right direction. Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the