Re: [asterisk-users] setting up phones
Hi Ott, Have you made it work with Asterisk and Aastra IP Phone. I am also trying the same thing, in Asterisk it shows registered OK but when I dial from extension to extension, call is failed... Please let me know have you made it work...:( On Mon, Jul 13, 2009 at 11:46 PM, Ott Rose sixfourimp...@hotmail.comwrote: I did set sip debug on from the CLI It doesn't scroll messages like it did on Fri i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which isn't either one of the ips of the asterisk server. then it hung up i do have a dial tone i just figured something out after reading my post. if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to the extension and the other phone rings. still can't get the 99 to call the asterisk server to work i put in the ips of the server but it hangs up right away -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:57:59 -0500 Subject: Re: [asterisk-users] setting up phones I assume you get a dial tone when you pick up the handset?If you had a good phone-to-asterisk connection, debug would show a connection or rejection when you did 99#. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Monday, July 13, 2009 12:49 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You’re really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and “functional” but that you couldn’t make your phones talk to it. I’m thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Monday, July 13, 2009 12:02 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Windows Live™ Hotmail®: Spread the word when you add celeb photos to your e-mails. Check it out.http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_WL_QA_HM_celebrity_photos1_072009cat=celebrity -- Bing™ brings you health information from trusted sources. Try it now.http://www.bing.com/search?q
Re: [asterisk-users] setting up phones
Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™ Hotmail®: Spread the word when you add celeb photos to your e-mails. Check it out. http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_WL_QA_HM_celebrity_photos1_072009cat=celebrity___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You're really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and functional but that you couldn't make your phones talk to it. I'm thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows LiveT HotmailR: Spread the word when you add celeb photos to your e-mails. Check it out. http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_ WL_QA_HM_celebrity_photos1_072009cat=celebrity ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You’re really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and “functional” but that you couldn’t make your phones talk to it. I’m thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Windows Live™ Hotmail®: Spread the word when you add celeb photos to your e-mails. Check it out. _ Bing™ brings you health information from trusted sources. Try it now. http://www.bing.com/search?q=pet+allergyform=MHEINApubl=WLHMTAGcrea=TXT_MHEINA_Health_Health_PetAllergy_1x1___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I assume you get a dial tone when you pick up the handset?If you had a good phone-to-asterisk connection, debug would show a connection or rejection when you did 99#. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You're really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and functional but that you couldn't make your phones talk to it. I'm thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows LiveT HotmailR: Spread the word when you add celeb photos to your e-mails. Check http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_ WL_QA_HM_celebrity_photos1_072009cat=celebrity it out. _ BingT brings you health information from trusted sources. Try it now. http://www.bing.com/search?q=pet+allergyform=MHEINApubl=WLHMTAGcrea=TXT_ MHEINA_Health_Health_PetAllergy_1x1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I did set sip debug on from the CLI It doesn't scroll messages like it did on Fri i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which isn't either one of the ips of the asterisk server. then it hung up i do have a dial tone i just figured something out after reading my post. if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to the extension and the other phone rings. still can't get the 99 to call the asterisk server to work i put in the ips of the server but it hangs up right away From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:57:59 -0500 Subject: Re: [asterisk-users] setting up phones I assume you get a dial tone when you pick up the handset?If you had a good phone-to-asterisk connection, debug would show a connection or rejection when you did 99#. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You’re really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and “functional” but that you couldn’t make your phones talk to it. I’m thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Windows Live™ Hotmail®: Spread the word when you add celeb photos to your e-mails. Check it out. Bing™ brings you health information from trusted sources. Try it now. _ Bing™ brings you health information from trusted sources. Try it now. http://www.bing.com/search?q=pet+allergyform=MHEINApubl=WLHMTAGcrea=TXT_MHEINA_Health_Health_PetAllergy_1x1___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
the “sip show peers command returns Name/username HostDyn Nat ACL Port Status 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] I ran grep on the sip.conf and it didn't find any IPs. Where would I add my IP? I am guessing that the phones will not work for eternal calling if my SIP trunk is not configured correctly. I have had trouble finding the correct settings for the SIP trunk. I am still learning the termanology which has been part of my problem. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 16:17:07 -0500 Subject: Re: [asterisk-users] setting up phones What do you get from “sip show peers” in CLI? Do you have your ip address in sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. Lauren found her dream laptop. Find the PC that’s right for you. Windows Live™: Keep your life in sync. Check it out. _ Lauren found her dream laptop. Find the PC that’s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290___ -- Bandwidth and Colocation Provided
Re: [asterisk-users] setting up phones
Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.comwrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Asterisk registers with the phones? Obviously I have zero experience with these sets, but that is a new one. Thanks, Steve On Fri, Jul 10, 2009 at 9:40 AM, Danny Nicholas da...@debsinc.com wrote: Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
My bad. Asterisk does not register with the phone. It can send out SIP headers to make the phones re-register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, July 10, 2009 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] setting up phones Asterisk registers with the phones? Obviously I have zero experience with these sets, but that is a new one. Thanks, Steve On Fri, Jul 10, 2009 at 9:40 AM, Danny Nicholas da...@debsinc.com wrote: Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Extension 500 is registered just fine. 200 OK Maybe you should start with a GUI version of Asterisk. Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? Thanks, Steve Totaro On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose sixfourimp...@hotmail.comwrote: Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog ' 5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060 ;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1 sip%3a...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 -- Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.comwrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.comwrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. _ Hotmail® has ever-growing storage! Don’t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I have the GUI setup and I setup users in the gui before. I still couldn't get it to work. I don't have any SIP trunks setup via the GUI because I can't figure out my settings and I was told I didn't need it to test extensions. I am not sure what you mean by Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? I don't know how to do that. Date: Fri, 10 Jul 2009 11:08:59 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Extension 500 is registered just fine. 200 OK Maybe you should start with a GUI version of Asterisk. Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? Thanks, Steve Totaro On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose sixfourimp...@hotmail.com wrote: Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can talk to it and register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 it out. _ HotmailR has ever-growing storage! Don't worry about storage limits. Check it out. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutori al_Storage_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I saw 127.0.0.2, never seen that before. Loopback that I have seen is 127.0.0.1. I always just bind to 0.0.0.0 since I have never really seen a point to binding to a specific IP. I guess if you are dual homed and don't want remote phones to work, but then you could just block that stuff in IPTables or whatever firewall. Thanks, Steve T BTW, what GUI? That was part of what I was asking when I said what flavor of Asterisk? On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas da...@debsinc.com wrote: You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 -- Hotmail® has ever-growing storage! Don’t worry about storage limits. Check it out.http://windowslive.com/Tutorial/Hotmail/Storage?ocid
Re: [asterisk-users] setting up phones
Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP
Re: [asterisk-users] setting up phones
Asterisk GUI-version : SVN-branch-2.0-r4962 Date: Fri, 10 Jul 2009 11:57:38 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I saw 127.0.0.2, never seen that before. Loopback that I have seen is 127.0.0.1. I always just bind to 0.0.0.0 since I have never really seen a point to binding to a specific IP. I guess if you are dual homed and don't want remote phones to work, but then you could just block that stuff in IPTables or whatever firewall. Thanks, Steve T BTW, what GUI? That was part of what I was asking when I said what flavor of Asterisk? On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas da...@debsinc.com wrote: You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. Hotmail® has ever-growing storage! Don’t worry about storage limits. Check it out
Re: [asterisk-users] setting up phones
Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.comwrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060 ;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52 sip%3aaster...@10.0.0.52 ;tag=as66b3ded8 To: sip:5...@10.0.0.52 sip%3a...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 sip%3aaster...@10.0.0.52 Accept: application/sdp Content-Length: 0 -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone
Re: [asterisk-users] setting up phones
Phone 1 has 500 in all of it's id's and connects to server 10.0.0.52? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can talk to it and register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch
Re: [asterisk-users] setting up phones
yes From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 11:20:29 -0500 Subject: Re: [asterisk-users] setting up phones Phone 1 has 500 in all of it’s id’s and connects to server 10.0.0.52? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/500 10.0.0.52 D 5060 OK (1 ms) 501/501 10.0.0.52 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions
Re: [asterisk-users] setting up phones
Here is my conf files. sip.conf [general] context=default port=5060 ; UDP port for Asterisk bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP srvlookup=yes ; Enable DNS SRV server [500] type=peer context=phones host=dynamic fromuser=500 secret=500 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 500:5...@10.0.0.52/500 defaultip=10.0.0.60 mailbox=1001 disallow=all allow=alaw [501] type=peer context=phones host=dynamic fromuser=501 secret=501 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 501:5...@10.0.0.52/501 defaultip=10.0.0.46 mailbox=1002 disallow=all allow=alaw == users.conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm Date: Fri, 10 Jul 2009 12:16:50 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.com wrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should
Re: [asterisk-users] setting up phones
Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. On Fri, Jul 10, 2009 at 12:38 PM, Ott Rose sixfourimp...@hotmail.comwrote: Here is my conf files. sip.conf [general] context=default port=5060 ; UDP port for Asterisk bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP srvlookup=yes ; Enable DNS SRV server [500] type=peer context=phones host=dynamic fromuser=500 secret=500 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 500:5...@10.0.0.52/500 defaultip=10.0.0.60 mailbox=1001 disallow=all allow=alaw [501] type=peer context=phones host=dynamic fromuser=501 secret=501 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 501:5...@10.0.0.52/501 defaultip=10.0.0.46 mailbox=1002 disallow=all allow=alaw == users.conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm -- Date: Fri, 10 Jul 2009 12:16:50 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.comwrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060 ;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still
Re: [asterisk-users] setting up phones
Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. _ Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. _ Insert movie times and more without leaving HotmailR. See how. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor ial_QuickAdd_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't use Asterisk-GUI but the general idea of a GUI is so you don't have to modify files by hand. You use the graphical user interface to generate the entries you need. If you are using a GUI then don't touch the files. Just download EVB (Easy Vox Box) and use the GUI. If you want to mess with the conf files then download the source and compile it for a vanilla, non-gui installation. Pick one or the other until you know what you are doing. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 4:11 PM, Ott Rose sixfourimp...@hotmail.com wrote: added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 2:39 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. -- Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. Insert movie times and more without leaving Hotmail®. See how. _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Date: Fri, 10 Jul 2009 16:19:52 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't use Asterisk-GUI but the general idea of a GUI is so you don't have to modify files by hand. You use the graphical user interface to generate the entries you need. If you are using a GUI then don't touch the files. Just download EVB (Easy Vox Box) and use the GUI. If you want to mess with the conf files then download the source and compile it for a vanilla, non-gui installation. Pick one or the other until you know what you are doing. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 4:11 PM, Ott Rose sixfourimp...@hotmail.com wrote: added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _ Lauren found her dream laptop. Find the PC that http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 's right for you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. Lauren found her dream laptop. Find the PC that’s right for you. _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
What do you get from sip show peers in CLI? Do you have your ip address in sip.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _ Lauren found her dream laptop. Find the http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 PC that's right for you. _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
