[asterisk-users] Portech MV-378 with Asterisk
Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
The latest Nokia phones come with a SIP client and I like them. On Wed, Nov 5, 2008 at 10:56 PM, Pedram M [EMAIL PROTECTED] wrote: Any recommendations on good wireless SIP phones? Thanks, Pedram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( On Sat, Nov 1, 2008 at 1:22 PM, Emmanuel Pascal Bruno [EMAIL PROTECTED]wrote: Oh ok, I knew it was something like that. I have tried many different settings on my router. I'll dig into it some more. Thanks On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis [EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this You likely have firewall issues since it appears that you are not receiving a response from the other end. Make sure you have *both* your SIP and RTP ports forwarded to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
I have tried that too with no results On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis [EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( Disabling firewalls is almost certainly going to ensure the problem persists. You need to ensure that all SIP and RTP ports are port-forwarded from your firewall to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call problems
I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
Oh ok, I knew it was something like that. I have tried many different settings on my router. I'll dig into it some more. Thanks On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis [EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this You likely have firewall issues since it appears that you are not receiving a response from the other end. Make sure you have *both* your SIP and RTP ports forwarded to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call problems
I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED][EMAIL PROTECTED]for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED][EMAIL PROTECTED]- no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
