RE: [asterisk-users] PortSip and Astericks new install
Thanks for you reply, it does not output anything on the console when i make the call. However i have turned on SIP Debug and get the following: -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK32333 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Expires: 150 Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5d41ffa1 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK41 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Authorization: Digest username=4289, realm=asterisk, nonce=5d41ffa1, uri=sip:192.168.1.1:5060, response=c3f43fd747f5ef51168bbfa2401b680b, algorithm=MD5 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 [Kserver1*CLI Expires: 150 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- [Kserver1*CLI 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.2.3:8099: OPTIONS sip:[EMAIL PROTECTED]:8099 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 17 Nov 2006 20:47:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 150 Contact: sip:[EMAIL PROTECTED]:8099;expires=150 Date: Fri, 17 Nov 2006 20:47:18 GMT Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099;tag=18467 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: PortSIP softphone 2.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER, UPDATE Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' [Kserver1*CLI -- SIP read from 192.168.2.3:8099: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK31747 From: sip:[EMAIL PROTECTED]:5060;tag=30024 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Subject: call Expires: 120 Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 325 v=0 o=- 2177823 2177823 IN IP4 192.168.2.3 s=PortSIP VOIP SDK 2.0 c=IN IP4 192.168.2.3 t=0 0 m=audio 51636 RTP/AVP 0 3 8 97 4 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000
RE: [asterisk-users] PortSip and Astericks new install
I have tried the configuration exactly the same of yours and it's still not working. What could be wrong with my installation? I first of all tried the packages in Debian stable, this didn't work so I compiled from source but still the problem occurs. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 16 November 2006 00:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: I just downloaded and installed PortSip2 I have an extension setup as [4289] type = friend host = dynamic username=4289 qualify=500 reinvite=no canreinvite=no nat=yes dtmfmode = rfc2833 context = managers callgroup=1 pickupgroup=1,2 mailbox = [EMAIL PROTECTED] secret=12345 disallow=all allow=ulaw allow=alaw allow=slin callerid = Doug Lytle 4289 PortSip2: Account: 4289 AuthName: 4289 Password: 12345 Server: YourAsteriskServer Port: 5060 When setup, it just worked. I'll have to guess that your Asterisk installation is a fault. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PortSip and Astericks new install
Charlie Grosvenor wrote: I have tried the configuration exactly the same of yours and it's still not working. What could be wrong with my installation? I first of all Show us what being displayed on the Asterisk console, also show the output from sip show peers. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
What do you mean by asterisk console, I ran asterisk -r and entered sip show peers and this is what i got: server1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 4289/4289 192.168.2.3 D N 8925 OK (2 ms) John (Unspecified)D 0Unmonitored 2 sip peers [2 online , 0 offline] This any good? Thanks From: [EMAIL PROTECTED] on behalf of Doug Lytle Sent: Thu 16/11/2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: I have tried the configuration exactly the same of yours and it's still not working. What could be wrong with my installation? I first of all Show us what being displayed on the Asterisk console, also show the output from sip show peers. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PortSip and Astericks new install
Charlie Grosvenor wrote: What do you mean by asterisk console, I ran asterisk -r and entered sip show peers and this is what i got: server1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 4289/4289 192.168.2.3 D N 8925 OK (2 ms) John (Unspecified)D 0Unmonitored 2 sip peers [2 online , 0 offline] Asterisk -r is reconnecting you to the console. This shows that your soft phone is registered. What does it show when you try to dial 500? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PortSip and Astericks new install
I have just installed Asterisk and installed the sample configuration files. Asterisks appears to be working and I have added a SIP client: [John] type=friend secret=test host=dynamic allow=all I have been trying to dial the demo number 500 when using PortSip, Asterisks answers the phone but PortSip gives me the error: Call failed: codec not accepted 488. I have tried changing the enabled codecs in PortSip but this makes no difference. I have also tried various other SoftPhone but none of them seem to work. Anybody know what I have missed / doing wrong? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PortSip and Astericks new install
try : [John] type=friend secret=test host=dynamic disallow=all allow =gsmilbculawalaw Also try other sip phone slike sjphone just to make sure there is no prob . On 16/11/06, Charlie Grosvenor [EMAIL PROTECTED] wrote: I have just installed Asterisk and installed the sample configuration files. Asterisks appears to be working and I have added a SIP client: [John] type=friend secret=test host=dynamic allow=all I have been trying to dial the demo number 500 when using PortSip, Asterisks answers the phone but PortSip gives me the error: Call failed: codec not accepted 488. I have tried changing the enabled codecs in PortSip but this makes no difference. I have also tried various other SoftPhone but none of them seem to work. Anybody know what I have missed / doing wrong? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PortSip and Astericks new install
Charlie Grosvenor wrote: [John] type=friend secret=test host=dynamic allow=all Try: disallow=all allow=alaw allow=ulaw allow=gsm Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: Initiated sip call to 500 Call has disconnected. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 15 November 2006 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: [John] type=friend secret=test host=dynamic allow=all Try: disallow=all allow=alaw allow=ulaw allow=gsm Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PortSip and Astericks new install
Charlie Grosvenor wrote: Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: I just downloaded and installed PortSip2 I have an extension setup as [4289] type = friend host = dynamic username=4289 qualify=500 reinvite=no canreinvite=no nat=yes dtmfmode = rfc2833 context = managers callgroup=1 pickupgroup=1,2 mailbox = [EMAIL PROTECTED] secret=12345 disallow=all allow=ulaw allow=alaw allow=slin callerid = Doug Lytle 4289 PortSip2: Account: 4289 AuthName: 4289 Password: 12345 Server: YourAsteriskServer Port: 5060 When setup, it just worked. I'll have to guess that your Asterisk installation is a fault. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
