Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: Well if you say it's registered, then packets are getting to asterisk and asterisk is accepting them, and you've allowed that SIP client. So... if you say there's absolutely NOTHING happening when the phone dials, then it sure seems like the phone is bad -- again, assuming no event whatsoever is happening when you dial. What else have you done to debug this? Have you registered the phone directly against another * box? Have you registered another phone against this * box? *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status 6002/6002 10.10.10.126 D N 255.255.255.255 1720 OK (58 ms) 6001/6001 10.10.10.125 D N 255.255.255.255 5061 OK (5 ms) 2 sip peers [2 online , 0 offline] if I dial from one phone to the other, sometimes it goes throu, sometimes the phone just keep silent and NOTHING is shown in the *CLI prompt Since it is from both phones, which work on another asterisk box, I can be sure, that the phones are working. The network is just two phones, one hub and to the asterisk box. home*CLI show version Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-06-28 04:53:43 bye Ronald ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! bye Ronald ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On Mon, Jul 04, 2005 at 05:19:39PM +0800, Ronald_Wiplinger wrote: Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! How do you know it is not connected? Why do you assume it should be connected? Please answer your questions, and while you do: verify all of your assumptions. After you've answered them, please try to guess what our next question would have been. Is there a sip peer or it in sip.conf? How does that sip peer appear in 'sip show peers' on the CLI? voip-info,org, google, and such are valueble resources for answering the questions. For example: connected basically means (for a SIP client) being registered as a SIP peer. Though a client can technically connect without registrating in advance. So: is there a section for it in sip.conf? How does it appear in the output of 'sip show peers'? Do you have any reason to believe that the grandstream phone is actually sending any packets to your asterisk computer? Try running: tcpdump -n 'host IP_ADDRESS_THE_PHONE' on your asterisk system. Ethereal may be useful for protocol analisys, but tcpdump is great if you just want to know if there is traffic. Naturally another thing to try is to eliminate one part of the problem: can you use a different SIP client with the same definitions of the server (or vice-versa)? Does that SIP client work with any other SIP server? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! Do you have qualify=1000 or some value in the sip.conf? Are you getting a time when you do a sip show peers? It could be the phone is registering and then losing network, and if the registration time is an hour it would still show as registered even if it was uncontactable. (I think). IANAAE (I am not an asterisk expert.) e.g. 212/ 192.168.0.25 D 255.255.255.255 5062 OK (24 ms) 211/ 192.168.0.25 D 255.255.255.255 5060 OK (27 ms) 210/ 192.168.0.23 D 255.255.255.255 5060 OK (59 ms) 203/ (Unspecified)D 255.255.255.255 0UNKNOWN Regards Mark ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! Well if you say it's registered, then packets are getting to asterisk and asterisk is accepting them, and you've allowed that SIP client. So... if you say there's absolutely NOTHING happening when the phone dials, then it sure seems like the phone is bad -- again, assuming no event whatsoever is happening when you dial. What else have you done to debug this? Have you registered the phone directly against another * box? Have you registered another phone against this * box? ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote: Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote: Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. bye Ronald ___ 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
[Asterisk-Users] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. What could be the reason? I have installed Festiva, and was only able once to listen a text to speech, since then this extension number never gives me a tone. Sometimes it shows up in the CLI, but without a tone on the phone. Other extensions have the same... bye Ronald ___ 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
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? -- Robert Goodyear Brand Up LLC http://www.brand-up.com ___ 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
