-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;******************************************* -And this is the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ****************************** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. -------------- next part -------------- A non-text attachment was scrubbed... Name: sip Type: application/octet-stream Size: 12847 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040329/0a958b8b/sip-0001.obj
What does your extensions.conf look like? Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of pesb Sent: 29 March 2004 18:48 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;******************************************* -And this is the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ****************************** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal.
On Monday 29 March 2004 12:25, pesb wrote:> I have 2 SIP GrandStream phones, both phones are correctly > registered to the Asterisk server. But, when I try to make a call > from registered phone '1005' to registered phone '1004', dialing > 1004, Asterisk responds with the 'Status: 404 Not Found' message.Have you thought about configuring extensions.conf ? -Tilghman
Try this small extensions.conf Don't think I have missed owt. My config files are here, you just need to add your own extension numbers. http://www.codepipe.com/id25.htm Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of pesb Sent: 29 March 2004 19:26 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;******************************************* -And this is the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ****************************** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. -------------- next part -------------- A non-text attachment was scrubbed... Name: extensions.conf Type: application/octet-stream Size: 684 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040329/308e97ff/extensions.obj
Robinson, Eliot S.
2004-Mar-29 12:58 UTC
[Asterisk-Users] Asterisk + GrandStream SIP phones
how do you get the phone message button to light when there is a message? eliot -----Original Message----- From: asterisk-users-admin@lists.digium.com on behalf of David J Carter Sent: Mon 3/29/2004 1:58 PM To: asterisk-users@lists.digium.com Cc: Subject: RE: [Asterisk-Users] Asterisk + GrandStream SIP phones -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3926 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040329/27915c7c/attachment.bin
Try to add a qualify=XXXX to sip.conf, and try to exec a sip show peers. In spite of phones appears like register, if you use NAT, your firewall can cut communication. Try the next: Just after phone register call to it, and then wait for a minutes and try to call again. Could you call first time but not in second one? It is due to your firewall. Try to configure wuth next config: [1004] ...... ..... qualify=XXXX ....... ...... In you grandstream configuration try to put time to expire register 1 minute and then try to do the previous test. I'm sorry for my english, but I hope this let you call. Regards, srsergio -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de pesb Enviado el: lunes, 29 de marzo de 2004 20:26 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secretreinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;******************************************* -And this is the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ****************************** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal.
Hi, Thanks for the help. You were correct. There was some data missing in the extension.conf file I was able to call one SIP phone from the other. I was even able to call an H323 IP phone registered to the gnugk GK (It has Asterisk registered to him as a GW). But, I have another problem rigth now. All the RTP Data Flow is passing through the Asterisk Proxy, which is a bad thing if I want to have many SIP phones in my system. How can I configure the SIP phone in order to make all RTP data flow directly from one SIP phone to the other? And, how can I configure it in order to make all RTP data flow directly from one SIP phone to the H323 IP phone (the one registered to my gnugk GK)? I would also like to be able to make calls from a SIP phone to the other SIP phone, but instead of having the ASTERISK PBX authorizing the calls, it would be the H323 GK the one that would authorize calls. How can I do this? Thanks again