tag:blogger.com,1999:blog-7262335442574749724.post5556026663043765499..comments2023-09-24T21:45:15.090+08:00Comments on SANJAY WILLIE'S Human Language. Asterisk | Nagios | OpenSource | Microsoft | Security: Asterisk 10 (1.10) SMS (messaging or SIP Messaging) in actionJayWShttp://www.blogger.com/profile/04318296929423691109noreply@blogger.comBlogger49125tag:blogger.com,1999:blog-7262335442574749724.post-9254707873014007802019-02-19T23:22:00.324+08:002019-02-19T23:22:00.324+08:00[Feb 19 20:43:48] ERROR[10799][C-00000001]: pbx_fu...[Feb 19 20:43:48] ERROR[10799][C-00000001]: pbx_functions.c:608 ast_func_read: Function CUT not registered<br /> <br />unable to clear this error can anyone please help me?Pradeepnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-35650189569436596372019-02-19T23:17:51.698+08:002019-02-19T23:17:51.698+08:00hi i am getting this error can you p[lease tell me... <br />hi i am getting this error can you p[lease tell me y this happening??<br /><br /><br />Executing [7002@msgcontext:11] Set("Message/ast_msg_queue", "ME_1=") in new stack<br />[Feb 19 20:43:48] ERROR[10799][C-00000001]: pbx_functions.c:608 ast_func_read: Function CUT not registered<br />pradeepnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-47305392965117591902014-06-30T23:38:19.903+08:002014-06-30T23:38:19.903+08:00Hi,
I'm running the FreePBX distro and cannot...Hi,<br /><br />I'm running the FreePBX distro and cannot get it to work. I think it is related to the deviceanduser setup from FreePBX which makes it possible to use multiple devices for one user. This is the relevant log when I sent a message:<br /><br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:1] NoOp("Message/ast_msg_queue", "SMS receiving dialplan invoked") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:2] NoOp("Message/ast_msg_queue", "To sip:3344@172.16.42.119") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:3] NoOp("Message/ast_msg_queue", "From "3" ") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:4] NoOp("Message/ast_msg_queue", "Body Test 2") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:5] Set("Message/ast_msg_queue", "ACTUALTO=sip:3344") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:6] MessageSend("Message/ast_msg_queue", "sip:3344,"3" ") in new stack<br />[2014-06-30 15:16:32] WARNING[2716][C-00000001] chan_sip.c: Purely numeric hostname (3344), and not a peer--rejecting!<br />[2014-06-30 15:16:32] WARNING[2716][C-00000001] chan_sip.c: Purely numeric hostname (3344), and not a peer--rejecting!<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:7] NoOp("Message/ast_msg_queue", "Send status is FAILURE") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:8] GotoIf("Message/ast_msg_queue", "1?sendfailedmsg") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Goto (astsms,3344,10)<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:10] Set("Message/ast_msg_queue", "MESSAGE(body)="[30062014-15:16:32] Your message to 3344 has failed. Retry later."") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:11] Set("Message/ast_msg_queue", "ME_1=sip:3@172.16.42.119>") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:12] Set("Message/ast_msg_queue", "ACTUALFROM=sip:3") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:13] MessageSend("Message/ast_msg_queue", "sip:3,ServiceCenter") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: -- Executing [3344@astsms:14] Hangup("Message/ast_msg_queue", "") in new stack<br />[2014-06-30 15:16:32] VERBOSE[2716][C-00000001] pbx.c: == Spawn extension (astsms, 3344, 14) exited non-zero on 'Message/ast_msg_queue'<br /><br />Any ideas how to get this working with this setup?<br /><br />FreePBX release 6.5<br />Asterisk 11.10.2<br />deviceanduser mode enabled in Advanced Settings.<br /><br />Thanks,<br />Chris<br /><br />Chrisnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-2810606237445075772014-05-17T19:00:25.818+08:002014-05-17T19:00:25.818+08:00Those want offline try this: http://highsecurity.b...Those want offline try this: http://highsecurity.blogspot.com/2013/01/asterisk-10-or-11-messaging-smssip.htmlJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-59158798917063550602014-05-17T17:17:58.039+08:002014-05-17T17:17:58.039+08:00Thanks for the great article worked like a charm. ...Thanks for the great article worked like a charm. I'm using Freepbx 5.211.65-12 and Asterisk 11.9. My question is I know you outlined how to store a failed message to then send it when the phone comes back on line. Have you implemented it? If so can you submit additional information to this guide on how to do so? If you have not, can I donate $50 for your time to do it.<br /><br />Thanks for the great work.mikeisflynoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-38286265242822719722013-08-14T13:34:05.032+08:002013-08-14T13:34:05.032+08:00can u plz mail me the procedure how to create exte...can u plz mail me the procedure how to create extensions in X-lite,register the IP address of asterisk server and how to send SMS to asterisk from X_lite SIP phoneAnonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-37144568757336405082013-07-06T02:30:57.303+08:002013-07-06T02:30:57.303+08:00Did I have to open some ports on my Firewall ?? I ...Did I have to open some ports on my Firewall ?? I try on my local network and works find, when Im outside i have a Message error "Service Unavailable (503)" i make calls without any problems.<br /><br />ThanksYair Magananoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-48423603535334925732013-06-19T22:41:34.711+08:002013-06-19T22:41:34.711+08:00Hi all,
I am trying to enable SIP SIMPLE communi...Hi all,<br /><br /><br />I am trying to enable SIP SIMPLE communication in my test environment.<br /><br />I have the following env :<br /><br />- one server (192.168.50.126) with Asterisk 11<br />- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143<br /><br />I successfully had a phone call between clients.<br /><br />I used the following link to enable SIMPLE messaging between my clients : http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html<br /><br />Both users managed to register.<br /><br />Adding verbose on the server, I have the following traces when I send the message "MESSAGE FROM ALICE TO BOB" from "demo-alice" to "demo-bob"<br /><br />http://paste.fedoraproject.org/19489/37158861/<br /><br />As you can see I succeed to have the message sent from alice to Asterisk.<br /><br />When the server is trying to transmitting, I see a 401 error message. According to this post (http://forums.digium.com/viewtopic.php?f=1&t=72814) the first 401 should be normal as authentication is requested. <br /><br />Afterwards the server emit 202 message.<br /><br />But "demo-bob" never receives a message.<br />I ran wireshark on server and client. It confirms that no message is sent from Asterisk to "demo-bob".<br /><br />Adding NoOp traces in sip-messages context I see that I never go into it while it works when adding traces in users context during a call.<br /><br />Could you please give me advice ?<br /><br /><br />Here are my extensions.conf and sip.conf according to the link I mentioned.<br />http://paste.fedoraproject.org/19626/16493741/<br />http://paste.fedoraproject.org/19627/49423137/<br /><br /><br />Thanks a lot,<br /><br />EloiAnonymoushttps://www.blogger.com/profile/06276225148305226127noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-14021377030777861122013-06-19T22:39:19.877+08:002013-06-19T22:39:19.877+08:00Hi all,
I am trying to enable SIP SIMPLE communi...Hi all,<br /><br /><br />I am trying to enable SIP SIMPLE communication in my test environment.<br /><br />I have the following env :<br /><br />- one server (192.168.50.126) with Asterisk 11<br />- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143<br /><br />I successfully had a phone call between clients.<br /><br /><br />Both users managed to register.<br /><br />Adding verbose on the server, I have the following traces when I send the message "MESSAGE FROM ALICE TO BOB" from "demo-alice" to "demo-bob"<br /><br />http://paste.fedoraproject.org/19489/37158861/<br /><br />As you can see I succeed to have the message sent from alice to Asterisk.<br /><br />When the server is trying to transmitting, I see a 401 error message. According to this post (http://forums.digium.com/viewtopic.php?f=1&t=72814) the first 401 should be normal as authentication is requested. <br /><br />Afterwards the server emit 202 message.<br /><br />But "demo-bob" never receives a message.<br />I ran wireshark on server and client. It confirms that no message is sent from Asterisk to "demo-bob".<br /><br />I never have the trace NoOp added in messages-sip context. However if I add it in users context and performed a call between alice and bob, it works.<br /><br />Could you please give me advice ?<br /><br /><br />Here are my extensions.conf and sip.conf according to the link I mentioned.<br />http://paste.fedoraproject.org/19626/16493741/<br />http://paste.fedoraproject.org/19627/49423137/<br /><br /><br />Thanks a lot,<br /><br />EloiAnonymoushttps://www.blogger.com/profile/06276225148305226127noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-73385897980291461822013-06-11T18:06:44.153+08:002013-06-11T18:06:44.153+08:00send text messages to india
TelCan offers so many ...<a href="https://www.telcan.com/Features_India_calling_card.aspx" rel="nofollow">send text messages to india</a><br />TelCan offers so many features in US to India calling including the ability to cheapest calls to india to any mobile phone. The best calling cards TO calls from uk to india .<br /><a href="https://www.telcan.com/Refer_A_Friend_Cheap_India_Calls.aspx" rel="nofollow">phone in india</a><br />QAhttps://www.blogger.com/profile/08676921299344919058noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-32811963114010414582013-05-17T17:20:58.131+08:002013-05-17T17:20:58.131+08:00Hi folks,
I just tried this between 2 Ekiga clien...Hi folks,<br /><br />I just tried this between 2 Ekiga clients and the Windows Ekiga doesn't send Authorization header for the MESSAGE packets while the linux version does.<br />Result: from the linux client you can send messages from the windows you cannot...<br />This is the header what's missing at the windows client:<br /><br />Authorization: Digest username="99", realm="asterisk", nonce="18863da3", uri="sip:98@192.168.1.2", algorithm=MD5, response="..."<br /><br /><br />Franknoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-61627586240458616632013-04-30T00:48:24.245+08:002013-04-30T00:48:24.245+08:00Asterisnnow 3.0 with Freepbx or Pbxinaflash
This ...Asterisnnow 3.0 with Freepbx or Pbxinaflash<br /><br />This works.<br />Just copy and paste.<br />Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-76429550206198369142013-03-03T23:48:24.374+08:002013-03-03T23:48:24.374+08:00Dear Sanjay,
I use softphone client Jitsi but sen...Dear Sanjay,<br /><br />I use softphone client Jitsi but send message is always disable<br /><br />But if i use 3CXPhone it is enable<br /><br />Please help<br /><br />Mochammad Effendihttps://www.blogger.com/profile/00052447885585169408noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-70163756554347952052013-02-10T22:28:34.034+08:002013-02-10T22:28:34.034+08:00Justin
Edit
/etc/asterisk/logger.conf
Add this l...Justin<br /><br />Edit<br />/etc/asterisk/logger.conf<br /><br />Add this line<br />full => notice,warning,error,debug,verbose<br /><br />Stop and restart asterisk, your full log will be there!JayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-3974296872392095762013-02-09T03:12:53.268+08:002013-02-09T03:12:53.268+08:00I'm not finding the "full" log file....I'm not finding the "full" log file. I have messages, queue_log and h323_log?<br /><br />I have verified that messages are enabled via CLI>logger show channels<br /><br />Here is the pastebin for messages.conf<br />http://pastebin.com/a92zL50gAnonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-46727942063773803142013-02-01T15:26:10.411+08:002013-02-01T15:26:10.411+08:00SIP/2.0 401 Unauthorized
That can be normal.
I ...SIP/2.0 401 Unauthorized<br /><br />That can be normal. <br /><br />I need the logs, otherwise, sip traces dont tell a thing...JayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-42988927287057921162013-02-01T00:54:24.373+08:002013-02-01T00:54:24.373+08:00I wasn't able to gather that file but I was ab...I wasn't able to gather that file but I was able to copy the CLI logs during the handoffs. I am able to login with SIP and initiate a phone call. <br /><br />If you look you will see the PBX responds with 401 Unauthorized when I am trying to send a message.<br /><br />This is the same message I receive when doing a PCAP capture.<br /><br />http://pastebin.com/q7DMb0tH<br /><br />I still feel like my configs are messed up somewhere? Thank you very much for your help.Anonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-39122103441744861842013-01-25T21:36:25.954+08:002013-01-25T21:36:25.954+08:00Justin
Can the two phones call each other to begi...Justin<br /><br />Can the two phones call each other to begin with?<br /><br />Check the logs by doing<br /><br />tail -f /var/log/asterisk/full<br /><br />Put that in pastebin tooJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-16744454365807771742013-01-25T00:07:23.835+08:002013-01-25T00:07:23.835+08:00Also here is a picture of a PCAP capture showing t...Also here is a picture of a PCAP capture showing the message sending to my server from handset 6000 to 6001 and the server rejecting<br />http://oi46.tinypic.com/28h0j2a.jpg<br /><br />I do not know how to view the logs on the PBX I can get to the CLI. I use the graphical interface for everything and text editor for making changes to .conf files.<br /><br />again thank you.Anonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-85094020227455459952013-01-24T23:59:44.754+08:002013-01-24T23:59:44.754+08:00Thank you for the response and interest.
Sip.con...Thank you for the response and interest. <br /><br />Sip.conf - http://pastebin.com/PSRnqNqL<br /><br />Extensions.conf - http://pastebin.com/2beREFg0Anonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-91215911509070901322013-01-24T10:30:15.064+08:002013-01-24T10:30:15.064+08:00Run
chmod +x astqueue.sh
You can run but it will...Run<br /><br />chmod +x astqueue.sh<br /><br />You can run but it will.prob do nothing muchJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-67278396317399523392013-01-24T09:25:11.809+08:002013-01-24T09:25:11.809+08:00Hi,
Can i manualy run the sh file?
I think that i...Hi,<br /><br />Can i manualy run the sh file?<br />I think that is not producing the file.<br /><br />I' trying ./astqueue.sh with the parameters and i'm getting permicion denied.<br /><br />Thank you.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-5681625883491742912013-01-24T00:35:31.173+08:002013-01-24T00:35:31.173+08:00Yeah, send your configs in pastebin also the log w...Yeah, send your configs in pastebin also the log when you try to sms. CHeersJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-88816646260613936472013-01-23T22:48:02.278+08:002013-01-23T22:48:02.278+08:00I have followed the steps above and still do not h...I have followed the steps above and still do not have SMS. However, when my phones are disconnected from the PBX but connected to the network they send SMS perfectly with no server. Once SIP registers I can make calls but no SMS. Any ideas? I can provide my .conf files.Anonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-91002123911045520272013-01-23T22:46:06.736+08:002013-01-23T22:46:06.736+08:00I cannot get this to work. I am using Ubuntu 12.04...I cannot get this to work. I am using Ubuntu 12.04 LTS with Asterisk 11. I performed step 1,2,and 3. Still no luck. What is funny is the handsets send text without the PBX, once SIP is registered I can make calls but SMS fails.Anonymoushttps://www.blogger.com/profile/14205673045369222818noreply@blogger.com