[asterisk-users] asterisk sip problem
Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sip problem
Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sip problem
They are on the same lan the adapter is registered sip show peers Name/username HostDyn Nat ACL Port Status sippyskypeuser/sippyskype 192.168.2.765070 OK (1 ms) 1000/1000 192.168.2.76 D 5061 OK (1 ms) freephonie-out/0950607456 212.27.52.5 N 5060 OK (766 ms) callcentric/17772962667204.11.192.34N 5080 OK (206 ms) the pap2t's IP is 192.168.2.205 and the IP of the asterisk box is 192.168.2.76 sip show registry HostUsername Refresh State Reg.Time freephonie.net:5060 0950601785 Registered Wed, 09 Jul 2008 10:12:44 callcentric.com:5080177729x 46 Registered Wed, 09 Jul 2008 10:13:29 I use line2 of my pap2t (line 1 is not enabled). Here is the conf : http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg On 7/9/08, MFH [EMAIL PROTECTED] wrote: Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sip problem
I don't see anything obvious right away other than have you confirmed that the phone is actually working? Can you get it to ring? With my Sipura adapters that use Linksys software I can view the call status in the Info section which if you have that panel might tell you if the adapter thinks a call is coming in. I just looked at my Info page with a call coming in and I can see the call state as Ringing and a bunch of other details. Call Status Call 1 State: Ringing Call 1 Tone:Ring - Hold Call 1 Encoder: G711u Call 1 Decoder: G711u Call 1 FAX: No Call 1 Type:[L1]Inbound Call 1 Remote Hold: No Call 1 Callback:No Call 1 Peer Name: UNAVAILABLE Call 1 Peer Phone: 1X Call 1 Duration: Call 1 Packets Sent:0 Call 1 Packets Recv:0 Call 1 Bytes Sent: 0 Call 1 Bytes Recv: 0 Call 1 Decode Latency: 0 ms Call 1 Jitter: 0 ms Call 1 Round Trip Delay:0 ms Call 1 Packets Lost:0 Call 1 Packet Error:0 Call 1 Mapped RTP Port: 16420 0 [EMAIL PROTECTED] wrote: They are on the same lan the adapter is registered sip show peers Name/username HostDyn Nat ACL Port Status sippyskypeuser/sippyskype 192.168.2.765070 OK (1 ms) 1000/1000 192.168.2.76 D 5061 OK (1 ms) freephonie-out/0950607456 212.27.52.5 N 5060 OK (766 ms) callcentric/17772962667204.11.192.34N 5080 OK (206 ms) the pap2t's IP is 192.168.2.205 and the IP of the asterisk box is 192.168.2.76 sip show registry HostUsername Refresh State Reg.Time freephonie.net:5060 0950601785 Registered Wed, 09 Jul 2008 10:12:44 callcentric.com:5080177729x 46 Registered Wed, 09 Jul 2008 10:13:29 I use line2 of my pap2t (line 1 is not enabled). Here is the conf : http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg On 7/9/08, MFH [EMAIL PROTECTED] wrote: Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
