Peter Raaijmaker
2005-Jul-10 10:16 UTC
[Asterisk-Users] Incoming calls from BudgetPhone.nl
(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not working. To me it all looks fine, no warnings or what so ever ? The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya ? Does anyone know what I?m doing wrong???? ? Thanks, Peter. ? ? ------------------------------- output of sip debug ------------------------------- ? 11 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: <sip:31717110342@budgetphone.nl>;tag=as5dc83db4 To: <sip:31717110342@budgetphone.nl> Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:31717110342@192.168.2.3> Event: registration Content-Length: 0 ? ? --- server*CLI> <-- SIP read from 81.23.228.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: <sip:31717110342@budgetphone.nl>;tag=as5dc83db4 To: <sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.247a Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 CSeq: 102 REGISTER WWW-Authenticate: Digest realm="budgetphone.nl", nonce="42d15009299d7652e8da589cee2723af4b6a96ca" Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 ? ? --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name budgetphone.nl 12 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: <sip:31717110342@budgetphone.nl>;tag=as7e56000d To: <sip:31717110342@budgetphone.nl> Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="31717110342", realm="budgetphone.nl", algorithm=MD5, uri="sip:budgetphone.nl", nonce="42d15009299d7652e8da589cee2723af4b6a96ca", response="cd69279e6a2512fd48d267ceea3394da", opaque="" Expires: 120 Contact: <sip:31717110342@192.168.2.3> Event: registration Content-Length: 0 ? ? --- server*CLI> <-- SIP read from 81.23.228.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: <sip:31717110342@budgetphone.nl>;tag=as7e56000d To: <sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0 Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 CSeq: 103 REGISTER Contact: <sip:31717110342@62.131.187.108:5060>;q=0.00;expires=120 Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 ? ? --- (9 headers 0 lines)--- Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound Registration: Expiry for budgetphone.nl is 120 sec (Scheduling reregistration in 105000 ms) Destroying call '26dfb15414601a871799536a3de1f776@127.0.0.1' server*CLI> <-- SIP read from 81.23.228.150:5060: INVITE sip:31717110342@62.131.187.108:5060 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:31717110342@81.23.228.150;ftag=as47419911;lr=on> Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: "0031172651375" <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 To: <sip:31717110342@budgetphone.nl> Contact: <sip:0031172651375@212.203.28.2> Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 10 Jul 2005 16:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 ? v=0 o=root 26318 26318 IN IP4 212.203.28.2 s=session c=IN IP4 81.23.228.139 t=0 0 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ? --- (15 headers 15 lines)--- Using INVITE request as basis request - 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl Sending to 81.23.228.150 : 5060 (NAT) Found peer '31717110342' Reliably Transmitting (NAT) to 81.23.228.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506 0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: "0031172651375" <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 To: <sip:31717110342@budgetphone.nl>;tag=as3f35655f Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:31717110342@192.168.2.3> Proxy-Authenticate: Digest realm="asterisk", nonce="555b996d" Content-Length: 0 ? ? --- Scheduling destruction of call '3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl' in 15000 ms server*CLI> <-- SIP read from 81.23.228.150:5060: ACK sip:31717110342@62.131.187.108:5060 SIP/2.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 From: "0031172651375" <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl To: <sip:31717110342@budgetphone.nl>;tag=as3f35655f CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14-5 (i386/linux)) Content-Length: 0 ? ? --- (8 headers 0 lines)--- Destroying call '3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl' server*CLI> ? ?
Long short, Maybe X-Ten has an stun relay setup and Asterisk doesn't? Rene Kluwen Chimit> (this time with subject....) > > Hello, > > I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. > When I dial my budgetphone nr on a PSTN KPN line it immediately gives a > busy > tone. > I tried X-lite, which worked perfect, so my modem (with nat) probably is > not > the problem. > I did a sip debug and got the following output. > Because I?m new to Asterisk I can?t get the error why this is not working. > To me it all looks fine, no warnings or what so ever> ? > The settings in sip.conf and extensions.conf are identical to those of > http://www.voip-info.org/tiki-index.php?page=Talkin2ya > ? > Does anyone know what I?m doing wrong???? > ? > Thanks, > Peter. > ? > ? > ------------------------------- > output of sip debug > ------------------------------- > ? > 11 headers, 0 lines > Reliably Transmitting (no NAT) to 81.23.228.150:5060: > REGISTER sip:budgetphone.nl SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc > From: <sip:31717110342@budgetphone.nl>;tag=as5dc83db4 > To: <sip:31717110342@budgetphone.nl> > Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 > CSeq: 102 REGISTER > User-Agent: Asterisk PBX > Expires: 120 > Contact: <sip:31717110342@192.168.2.3> > Event: registration > Content-Length: 0 > ? > ? > --- > server*CLI> > <-- SIP read from 81.23.228.150:5060: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc > From: <sip:31717110342@budgetphone.nl>;tag=as5dc83db4 > To: > <sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.247a > Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 > CSeq: 102 REGISTER > WWW-Authenticate: Digest realm="budgetphone.nl", > nonce="42d15009299d7652e8da589cee2723af4b6a96ca" > Server: Sip EXpress router (0.8.14-5 (i386/linux)) > Content-Length: 0 > ? > ? > --- (9 headers 0 lines)--- > Responding to challenge, registration to domain/host name budgetphone.nl > 12 headers, 0 lines > Reliably Transmitting (no NAT) to 81.23.228.150:5060: > REGISTER sip:budgetphone.nl SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e > From: <sip:31717110342@budgetphone.nl>;tag=as7e56000d > To: <sip:31717110342@budgetphone.nl> > Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 > CSeq: 103 REGISTER > User-Agent: Asterisk PBX > Authorization: Digest username="31717110342", realm="budgetphone.nl", > algorithm=MD5, uri="sip:budgetphone.nl", > nonce="42d15009299d7652e8da589cee2723af4b6a96ca", > response="cd69279e6a2512fd48d267ceea3394da", opaque="" > Expires: 120 > Contact: <sip:31717110342@192.168.2.3> > Event: registration > Content-Length: 0 > ? > ? > --- > server*CLI> > <-- SIP read from 81.23.228.150:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e > From: <sip:31717110342@budgetphone.nl>;tag=as7e56000d > To: > <sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0 > Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1 > CSeq: 103 REGISTER > Contact: <sip:31717110342@62.131.187.108:5060>;q=0.00;expires=120 > Server: Sip EXpress router (0.8.14-5 (i386/linux)) > Content-Length: 0 > ? > ? > --- (9 headers 0 lines)--- > Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound > Registration: Expiry for budgetphone.nl is 120 sec (Scheduling > reregistration in 105000 ms) > Destroying call '26dfb15414601a871799536a3de1f776@127.0.0.1' > server*CLI> > <-- SIP read from 81.23.228.150:5060: > INVITE sip:31717110342@62.131.187.108:5060 SIP/2.0 > Max-Forwards: 10 > Record-Route: <sip:31717110342@81.23.228.150;ftag=as47419911;lr=on> > Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 > Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa > From: "0031172651375" > <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 > To: <sip:31717110342@budgetphone.nl> > Contact: <sip:0031172651375@212.203.28.2> > Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Date: Sun, 10 Jul 2005 16:37:54 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 345 > ? > v=0 > o=root 26318 26318 IN IP4 212.203.28.2 > s=session > c=IN IP4 81.23.228.139 > t=0 0 > m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:5 DVI4/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ? > --- (15 headers 15 lines)--- > Using INVITE request as basis request - > 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl > Sending to 81.23.228.150 : 5060 (NAT) > Found peer '31717110342' > Reliably Transmitting (NAT) to 81.23.228.150:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506 > 0 > Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa > From: "0031172651375" > <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 > To: <sip:31717110342@budgetphone.nl>;tag=as3f35655f > Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:31717110342@192.168.2.3> > Proxy-Authenticate: Digest realm="asterisk", nonce="555b996d" > Content-Length: 0 > ? > ? > --- > Scheduling destruction of call > '3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl' in 15000 ms > server*CLI> > <-- SIP read from 81.23.228.150:5060: > ACK sip:31717110342@62.131.187.108:5060 SIP/2.0 > Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 > From: "0031172651375" > <sip:0031172651375@voipgw01.budgetphone.nl>;tag=as47419911 > Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl > To: <sip:31717110342@budgetphone.nl>;tag=as3f35655f > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.8.14-5 (i386/linux)) > Content-Length: 0 > ? > ? > --- (8 headers 0 lines)--- > Destroying call '3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl' > server*CLI> > ? > ? > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Michiel van Baak
2005-Jul-10 10:34 UTC
[Asterisk-Users] Incoming calls from BudgetPhone.nl
On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote:> (this time with subject....) > > Hello, > > I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. > When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy > tone. > I tried X-lite, which worked perfect, so my modem (with nat) probably is not > the problem. > I did a sip debug and got the following output. > Because I?m new to Asterisk I can?t get the error why this is not working. > To me it all looks fine, no warnings or what so ever? > ? > The settings in sip.conf and extensions.conf are identical to those of > http://www.voip-info.org/tiki-index.php?page=Talkin2ya > ? > Does anyone know what I?m doing wrong???? > ?Can you show us the relevant part in sip.conf and extensions.conf. It is working fine here (cept for audio quality and stability of the sip registration, I'm trashing them soon) If you post it I can compare it with my setup and maybe that will show us what's going wrong on your setup -- Michiel van Baak http://michiel.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"
Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host=xxx.xxx.xxx.xx directive. That helped me receiving calls from my sip provider, which had exactly the same problem. Julian. On 7/10/05, Peter Raaijmaker <voip@boumakers.nl> wrote:> (this time with subject....) > > Hello, > > I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. > When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy > tone. > I tried X-lite, which worked perfect, so my modem (with nat) probably is not > the problem. > I did a sip debug and got the following output. > Because I'm new to Asterisk I can't get the error why this is not working. > To me it all looks fine, no warnings or what so ever? > > The settings in sip.conf and extensions.conf are identical to those of > http://www.voip-info.org/tiki-index.php?page=Talkin2ya > > Does anyone know what I'm doing wrong???? > > Thanks, > Peter. >