Re: [asterisk-users] Re: calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)

address) Allan Kamau wrote: Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic chan_sip.c set_destination: can't find address for host I have set

[asterisk-users] calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic chan_sip.c set_destination: can't find address for host I have set the canreinvite=no in the

[asterisk-users] Invite issues

Hi all, I am receiving handle_request_invite: Failed to authenticate errors on two VoIP gateway devices connected to my asterisks SIP server. The problem seems to be in my configuration. I will only focus on one of this devices in this mail. On this device I receive the error Sep 21 12:02:23

[asterisk-users] SPA-3102 PSTN-VoIP Gateway (quest for one stage dialing)

I have read in the manual the SPA-3102 does not support one stage dialing for PSTN-VoIP calls, this is indeed frustrating, is there a way to provide seamless PSTN-VoIP gateway calls. I finally managed to get the SPA-3102 to work this is done by connecting via the ethernet port named Internet, I

[asterisk-users] SPA 3102 does not even attempt to register

I don't see a -- Saved useragent line for the SPA 3102 device am trying to connect to Asterisk. I have similar configuration for the SPA 3102 as I have for another hard phone in the sip.conf file but the device (SPA 3102) does not even attempt to register. I have configured the device to

[asterisk-users] inbound call from GSM gateway: handle_request_invite: Failed to authenticate user

Hi all, I am getting a handle_request_invite: Failed to authenticate user error when I attempt to receive calls from a GSM gateway (I can successfully call through the device VoIP-GSM from asterisk). I have looked for a solution to this error but most point me to adding a register line which I've

[asterisk-users] running agi application in the background

I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. When the call is answered, the application checks to see the number of seconds (talk

[asterisk-users] Dialing out using SIP terminal

Hi all, I have a VoIP GSM (SIP) terminal that I have successfully configured and registered in asterisk, would like to: a)Answer calls via asterisk coming from this terminal. b)Route outbound calls to this terminal. What Dial command do I use so as to have the sip terminal dial an outside line,

Re: [Asterisk-Users] GSM gateway hardware

2 to 4 channels to start with. Allan. --- hakem voip [EMAIL PROTECTED] wrote: How many channels do you need per gateway ? I might have slution for you voip2gsm Regards On 7/20/05, Allan Kamau [EMAIL PROTECTED] wrote: Thanks Roger, I find the second option more interesting

[Asterisk-Users] GSM gateway hardware

Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been

Re: [Asterisk-Users] GSM gateway hardware

Thanks Roger, I find the second option more interesting, let me know once you've managed to provide asterisk support for the GSM modem. Allan. --- Roger Schreiter [EMAIL PROTECTED] wrote: Allan Kamau schrieb: ... I am looking for a GSM VoIP gateway for use with Hi, do you think