Re: [asterisk-users] SendText causes Retransmission errors
On 03/20/2012 01:08 PM, Matt Hamilton wrote: Date: Mon, 19 Mar 2012 10:31:52 -0500 From: kpflem...@digium.com 502 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060 Why did Asterisk CANCEL the call here? I assume it's part of the SLA implementation. As I mentioned in my original email, I'm using SendText to send a text message when the user picks up a line in a SLA setup. In this case, ext 124 is calling 104, and one of the lines on 104 is picking it up. Asterisk is connecting to that line and cancelling the first request?? (just guessing) same = n,SendText(hi) same = n,SLAStation(4*104_line104) *503 (for 493) 10.0.1.57 10.0.1.103 Status: 200 OK* 524 (503) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 This appears to be broken. The listing here claims this ACK is in response to the '200 OK' in packet 503, which itself was a final response to the MESSAGE request in packet 493. However, MESSAGE requests do not use ACK for a three-way handshake like INVITE requests do. In addition, this packet is going the wrong direction to be an ACK for packet 503, since it's going the same direction as packet 503 did. I use Wireshark to capture the packets, and Wireshark is reporting it that way; i.e. saying that Request Frame for the ACK is the OK (for MESSAGE). I guess it's incorrect. The order and direction of messages I posted in my previous email are taken directly from Wireshark. Frame 15 is MESSAGE Frame 19 is OK (for MESSAGE) Frame 20 is ACK (Wireshark is saying the Request Frame is 20 ??) I tried to post the full SIP capture here, but it got rejected because of the size of the post (about 280k). Yep, that's a lot. The next step is probably to open an issue in our issue tracker and upload the capture file there (feel free to compress it first to save time and space). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText causes Retransmission errors
On 03/18/2012 08:07 PM, Matt Hamilton wrote: Kevin,thanks for your response. Here is the more detailed Wireshark capture of the SIP traffic between phone (10.0.1.57) and Asterisk (10.0.1.103). The numbers between parentheses are Request Frames. I don't see an ACK for the 200 OK to the INVITE (491) for the dialplan that gives us Retransmission errors (without WAIT), but there is also no ACK for the same INVITE for the dialplan that works (with WAIT). If you still want to take a look at the full packet capture, I'll post it. Matt - Without WAIT(1) - we get Retransmisson errors 486 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 487 10.0.1.103 10.0.1.57 Status: 401 Unauthorized 490 (486) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 491 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 492 10.0.1.103 10.0.1.57 Status: 100 Trying 493 10.0.1.103 10.0.1.57 Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) *500 (for 491) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP* 501 10.0.1.103 10.0.1.57 Request: NOTIFY sip:104@10.0.1.57:5060 502 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060 Why did Asterisk CANCEL the call here? *503 (for 493) 10.0.1.57 10.0.1.103 Status: 200 OK* 524 (503) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 This appears to be broken. The listing here claims this ACK is in response to the '200 OK' in packet 503, which itself was a final response to the MESSAGE request in packet 493. However, MESSAGE requests do not use ACK for a three-way handshake like INVITE requests do. In addition, this packet is going the wrong direction to be an ACK for packet 503, since it's going the same direction as packet 503 did. Whatever method you used to generate this report seems to be broken. I can't tell you exactly how it is broken without seeing the headers in the messages, though. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText causes Retransmission errors
Kevin, thanks for your response. Here is the more detailed Wireshark capture of the SIP traffic between phone (10.0.1.57) and Asterisk (10.0.1.103). The numbers between parentheses are Request Frames. I don't see an ACK for the 200 OK to the INVITE (491) for the dialplan that gives us Retransmission errors (without WAIT), but there is also no ACK for the same INVITE for the dialplan that works (with WAIT). If you still want to take a look at the full packet capture, I'll post it. Matt - Without WAIT(1) - we get Retransmisson errors 48610.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 48710.0.1.103 10.0.1.57Status: 401 Unauthorized 490 (486) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 49110.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 49210.0.1.103 10.0.1.57Status: 100 Trying 49310.0.1.103 10.0.1.57Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) 500 (for 491) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP 50110.0.1.103 10.0.1.57Request: NOTIFY sip:104@10.0.1.57:5060 50210.0.1.103 10.0.1.57Request: CANCEL sip:104@10.0.1.57:5060 503 (for 493) 10.0.1.5710.0.1.103 Status: 200 OK 524 (503) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 525 (501) 10.0.1.5710.0.1.103 Status: 200 OK 52610.0.1.5710.0.1.103 Status: 487 Request Terminated 527 (for 502) 10.0.1.5710.0.1.103 Status: 200 OK 528 (502) 10.0.1.103 10.0.1.57Request: ACK sip:104@10.0.1.57:5060 585 (524) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 588 (524) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 803 (588) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 806 (588) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 1223 (806) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 1229 (806) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 2042 (1229) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 204410.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 288610.0.1.103 10.0.1.57Status: 200 OK, with SDP 288810.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 375210.0.1.103 10.0.1.57Status: 200 OK, with SDP 375510.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 - with WAIT(1). There is no more messages beyond 672 until the call is over. Everything is normal. There is no ACK for the OK for INVITE in 430 here either. 42510.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 42610.0.1.103 10.0.1.57Status: 401 Unauthorized 429 (425) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 43010.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 43110.0.1.103 10.0.1.57Status: 100 Trying 43210.0.1.103 10.0.1.57Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) 443 (for 432) 10.0.1.5710.0.1.103 Status: 200 OK 645 (for 430) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP 64610.0.1.103 10.0.1.57Request: NOTIFY sip:104@10.0.1.57:5060 64710.0.1.103 10.0.1.57Request: CANCEL sip:104@10.0.1.57:5060 667 (443) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 668 (646) 10.0.1.5710.0.1.103 Status: 200 OK 67010.0.1.5710.0.1.103 Status: 487 Request Terminated 671 (647) 10.0.1.5710.0.1.103 Status: 200 OK 672 (for 647) 10.0.1.103 10.0.1.57Request: ACK sip:104@10.0.1.57:5060 Date: Fri, 16 Mar 2012 10:22:49 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText causes Retransmission errors On 03/16/2012 09:43 AM, Matt Hamilton wrote: Hi
[asterisk-users] SendText causes Retransmission errors
Hi, I'm using SendText to send a text message when the user picks up a line in a SLA setup (even though I'm not sure the problem is related to SLA). I'm on Asterisk 10.2.1 (same in 1.8.9) [from-office] .. same = n,SendText(hi) same = n,SLAStation(line1234) .. Here is a simplified version of the SIP messages: 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 Asterisk = phone OK (for the INVITE at 1) 5 phone = Asterisk OK (for the MESSAGE at 3) 6 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 7 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 .. The text message is sent and the call is connected, but Asterisk keeps resending OK for the INVITE, and eventually drops the call after Transmission timeout. If I insert a WAIT after SendText, the order of the OKs changes, and everything works: same = n,SendText(hi) same = n,Wait(1) same = n,SLAStation(line1234) Here is the SIP message flow with WAIT (4 and 5 above are swapped): 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 phone = Asterisk OK (for the MESSAGE at 3) 5 Asterisk = phone OK (for the INVITE at 1) Is there anything else I can do other than using WAIT (which might not be a consistent solution anyway)? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText causes Retransmission errors
On 03/16/2012 09:43 AM, Matt Hamilton wrote: Hi, I'm using SendText to send a text message when the user picks up a line in a SLA setup (even though I'm not sure the problem is related to SLA). I'm on Asterisk 10.2.1 (same in 1.8.9) [from-office] .. same = n,SendText(hi) same = n,SLAStation(line1234) .. Here is a simplified version of the SIP messages: 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 Asterisk = phone OK (for the INVITE at 1) 5 phone = Asterisk OK (for the MESSAGE at 3) 6 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 7 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 .. Did the phone send an ACK for message 4? If not, that explains why Asterisk is retransmitting the '200 OK'. Posting a packet capture of this problem occurring would probably provide the details necessary to figure out what is going on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendtext() SIP MESSAGE to Bria or Eyebeam
Hello! I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria or a recent Eyebeam on my mac. I know it used to work, but right now I get 100 trying and nothing else from the softphone. Anyone that knows what's going on here? Thanks, /O -- _ -- 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] SendText
i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 7: Unclutter your desktop. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___ -- 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] SendText
OK. Thanks On Thu, Nov 12, 2009 at 4:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.comwrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 7: Unclutter your desktop. Learn more.http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009 ___ -- 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 ___ -- 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
[asterisk-users] SendText
Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). ___ -- 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] SendText
On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] SendText
Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 ___ -- 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] SendText
On 10/11/09 4:19 PM, Thomas Perron wrote: Will text messages work to non-SIP enpoints using your logic/code? thank you If you mean SMS, yeah. Basically use SendText for devices which can display them (i.e. SIP/IAX phones) and Clickatel or the like for disconnected devices (i.e. SMS to mobile). If you wanted to extend it you could also use the Jabber functions to send to instant messaging clients. Here at the offices we basically do the following: SMS Messages for urgent notifications, payments received, support requests. Jabber Messages for incoming support call details, long Post Dial Delay warnings, congestion warnings. MRTG displaying IAX2 and SIP peer response times. Custom graphs to display inter country links. We use a system of circles around an international link. Each of our servers gets a circle. The larger the circle, the higher the delay, and if the host is unreachable the circle goes red. That way you can see from a quick glance if an international link is totally down (lots of red circles), a problem for one of our servers (one red circle), or if one of our servers is having trouble connecting to all remote links (one red circle on each link). We do the same circles for a couple of key customers to make sure their systems are always connected to multiple of our exchanges. Oh, the other thing we display on the dashboard is our Jabber statuses, and the number of tickets open in any of our support queues, and who they are assigned to. That way if someone is getting overloaded with support requests you can move jobs to another staff member. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
[asterisk-users] SendText and sipsak
Hi, Following advice in voip-info.org, I could successfully send text to a remote SIP endpoint using sipsak and this command : # sipsak -M -v -s sip:7...@192.168.100.123 sip%3a7...@192.168.100.123 -B Lunch time warning: ignoring -i option when in usrloc mode timeout after 500 ms timeout after 1000 ms timeout after 2000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms *** giving up, no final response after 35621.047 ms Is normal for an endpoint to display a SIP MESSAGE without acking it ? Is there a better way to send a text to a remote end without sipsak ? I tried using .call file but couldn't set autoanswer. Regards ___ -- 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
[asterisk-users] SendText and non-ASCII characters
Hi, Is is possible to translate non-english text into ASCII text so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. Regards ___ -- 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] SendText and non-ASCII characters
Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Many programming languages can do that out of the box or there are extensions or libraries available. so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. You could try to add a charset attribute like so: Content-Type: text/plain; charset=utf-8 but it's unlikely that any phones pay attention. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] SendText and non-ASCII characters
Philipp Kempgen schrieb: Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Forgot to add some pointers. http://en.wikipedia.org/wiki/Unicode_normalization http://www.unicode.org/unicode/faq/normalization.html Many programming languages can do that out of the box or there are extensions or libraries available. http://www.php.net/manual/en/book.unicode.php http://www.php.net/manual/en/book.recode.php http://www.php.net/manual/en/book.iconv.php http://www.icu-project.org/ ... Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] SendText and non-ASCII characters
Philipp Kempgen schrieb: Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Many programming languages can do that out of the box or there are extensions or libraries available. https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/lib/utf8-normalize/ If your non-english text is in UTF-8 encoding the gs_utf8_decompose_to_ascii() function in gs_utf_normal.php does what you need. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] SendText and non-ASCII characters
Hi, At the moment, I'm trying to send Unicoded text to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? regards ___ -- 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] SendText and non-ASCII characters
Olivier schrieb: At the moment, I'm trying to send Unicoded text Unicode is not an encoding. It's just a list or table of characters (glyphs). http://en.wikipedia.org/wiki/Unicode Unicode is typically represented in encodings (misleadingly called charsets) such as UTF-8, UTF-16 ... http://en.wikipedia.org/wiki/UTF-8 http://en.wikipedia.org/wiki/UTF-16 to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? Interesting. Maybe an Asterisk developer can comment on that. I'd try to type the character (latin small letter e with acute) in the text editor of your choice and either save the file in ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a hexdump (hd) it has 2 bytes: C3 A9 http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233 But I thought you were trying to avoid non-english characters because the phone doesn't display them anyway. If that's what you want then just send one of the decompositioned forms, namely e´ or just e (easy to type). Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] SendText and non-ASCII characters
2008/11/24 Philipp Kempgen [EMAIL PROTECTED] Olivier schrieb: At the moment, I'm trying to send Unicoded text Unicode is not an encoding. It's just a list or table of characters (glyphs). http://en.wikipedia.org/wiki/Unicode Unicode is typically represented in encodings (misleadingly called charsets) such as UTF-8, UTF-16 ... http://en.wikipedia.org/wiki/UTF-8 http://en.wikipedia.org/wiki/UTF-16 to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? Interesting. Maybe an Asterisk developer can comment on that. I'd try to type the character (latin small letter e with acute) in the text editor of your choice and either save the file in ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a hexdump (hd) it has 2 bytes: C3 A9 http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233 But I thought you were trying to avoid non-english characters because the phone doesn't display them anyway. Obviously, the phone (Thomson st2030s) displays several latin charsets but the media to use for that is to use SIP MESSAGE. Thanks to your (crystal clear) explaination, I suppose I can't tailor SendText to use UTF-8 encoding so I typed the decompositioned form (ie e´). It doesn't display the way I wanted to. If I could simply use non-ascii in dialplay functions ... I also tried URIENCODE ... If that's what you want then just send one of the decompositioned forms, namely e´ or just e (easy to type). Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 ___ -- 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] SendText and non-ASCII characters
Olivier schrieb:
2008/11/24 Philipp Kempgen [EMAIL PROTECTED]
Olivier schrieb:
At the moment, I'm trying to send Unicoded text
Unicode is not an encoding. It's just a list or table of characters
(glyphs).
http://en.wikipedia.org/wiki/Unicode
Unicode is typically represented in encodings (misleadingly called
charsets) such as UTF-8, UTF-16 ...
http://en.wikipedia.org/wiki/UTF-8
http://en.wikipedia.org/wiki/UTF-16
to a SIP phone using
dialplan application SendText.
SendText(Hello World) works.
How can I insert letter 00E9 (from
http://www.unicode.org/charts/PDF/U0080.pdf) which can be written
eacute;
in HTML ?
Interesting. Maybe an Asterisk developer can comment on that.
I'd try to type the character (latin small letter e with acute) in
the text editor of your choice and either save the file in
ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a
hexdump (hd) it has 2 bytes: C3 A9
http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233
But I thought you were trying to avoid non-english characters
because the phone doesn't display them anyway.
Obviously, the phone (Thomson st2030s) displays several latin charsets but
the media to use for that is to use SIP MESSAGE.
Thanks to your (crystal clear) explaination, I suppose I can't tailor
SendText to use UTF-8 encoding so I typed the decompositioned form (ie
e´).
It doesn't display the way I wanted to.
If I could simply use non-ascii in dialplay functions ...
The required modification to add ;charset=UTF-8 to the Content-
Type header is simple and has already been done in Asterisk 1.6.1
(not in 1.6.0). It's in the add_text() function in chan_sip.c:
http://svn.digium.com/view/asterisk/tags/1.4.22/channels/chan_sip.c?view=markup#l_6229
http://svn.digium.com/view/asterisk/tags/1.6.0.1/channels/chan_sip.c?view=markup#l_7747
http://svn.digium.com/view/asterisk/tags/1.6.1-beta1/channels/chan_sip.c?view=markup#l_8022
/*! \brief Add text body to SIP message */
static int add_text(struct sip_request *req, const char *text)
{
/* XXX Convert \n's to \r\n's XXX */
- add_header(req, Content-Type, text/plain);
+ add_header(req, Content-Type, text/plain;charset=UTF-8);
add_header_contentLength(req, strlen(text));
add_line(req, text);
return 0;
}
You could easily make the same modification in 1.4 or 1.6.0.
It may help or it may not. Depends on the phone.
I also tried URIENCODE ...
Not the way to go here.
If that's what you want then just send one of the decompositioned
forms, namely e´ or just e (easy to type).
Philipp Kempgen
--
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. Tell that to my cheap analog caller id phone :) BTW, the sample zapata.conf in Asterisk also have the caller id names quoted. Maybe Mark can enlighten us :) Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says unknown or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Can post your zapata.conf? You need to ensure Asterisk is sending the FSK signal at the right time. This is from my zapata.conf: signalling=fxo_ks sendcalleridafter=2 usecallerid=yes cidsignalling=bell cidstart=ring Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. Tell that to my cheap analog caller id phone :) BTW, the sample zapata.conf in Asterisk also have the caller id names quoted. Maybe Mark can enlighten us :) Since the telco never sends quotes on PSTN calls, I can't imagine how this could be the case. Remember, in most cases, quotes in Asterisk config files are considered part of the value. So if you did a callerid=Robert Dobbs 5556661212 and then did a Noop($CALLERIDNAME) you would see Robert Dobbs on the Asterisk CLI. Maybe 1.2 and later silently strip off the quotes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Leo Ann Boon [EMAIL PROTECTED] Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says unknown or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Can post your zapata.conf? Forgot to mention, I was referring to calling from Asterisk itself (like using console) to another Asterisk via SIP without registration. (Same as the original post.) Unlike a soft or hard SIP phone, Asterisk's sip.conf has only one parameter callerid. Yuan Liu You need to ensure Asterisk is sending the FSK signal at the right time. This is from my zapata.conf: ... Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText() question
I have an F3000 phone utstarcom and sending a text message to it. All is working but there is a line of sender: asterisk. How do I control what this line says? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SendText() question
From: Jerry Geis [EMAIL PROTECTED] I have an F3000 phone utstarcom and sending a text message to it. All is working but there is a line of sender: asterisk. How do I control what this line says? Try sip.conf, callerid=... Yuan Liu THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText() question
It needed BOTH the text callerid and numerid callerid to display the text form. Callerid: Some name number At first I was only supplying the name. Works fine. Thanks, Jerry /From: Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users // //I have an F3000 phone utstarcom and sending a text message to it. //All is working but there is a line of sender: asterisk. //How do I control what this line says? / Try sip.conf, callerid=... Yuan Liu /THanks, // //Jerry/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Jerry Geis [EMAIL PROTECTED] It needed BOTH the text callerid and numerid callerid to display the text form. Callerid: Some name number Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Yuan Liu At first I was only supplying the name. Works fine. Thanks, Jerry /From: Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users // //I have an F3000 phone utstarcom and sending a text message to it. //All is working but there is a line of sender: asterisk. //How do I control what this line says? / Try sip.conf, callerid=... Yuan Liu /THanks, // //Jerry/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says unknown or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText Queue Notification
Hi, we have few cisco's...is there a way to push the queue information to the phone ?thanks in advance,jean-louis2006/8/24, Brodie Macleod [EMAIL PROTECTED]:I know this isn't answering your question, but what I did for queue notification was use softkeys on the phones that call a PHP script on the *box that'll output XML for the phone to parse and display the queue stats ondemand. Of course your phone would need to have an XML parser or some other type of minibrowser.For sending SIP messages to my Snom phones I use Sipsakto display agent login info and their associated queue(s) so that it's easyfor agents to know what their status is.-Brodie On Thursday 24 August 2006 10:33 am, John D. Coleman wrote: I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue.The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the queue.I tried using ChanSpy, but then SendText will send messages only to and from the caller who initiated the ChanSpy. One way I could get around this is if I found out how to use SendText from the commandline, like smsq. I'm pretty sure that's not possible because of the nature of SIP MESSAGE but I figured I'd ask. Thanks, John Coleman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText Queue Notification
I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue. The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the queue. I tried using ChanSpy, but then SendText will send messages only to and from the caller who initiated the ChanSpy. One way I could get around this is if I found out how to use SendText from the commandline, like smsq. I'm pretty sure that's not possible because of the nature of SIP MESSAGE but I figured I'd ask. Thanks, John Coleman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText Queue Notification
I know this isn't answering your question, but what I did for queue notification was use softkeys on the phones that call a PHP script on the * box that'll output XML for the phone to parse and display the queue stats on demand. Of course your phone would need to have an XML parser or some other type of minibrowser. For sending SIP messages to my Snom phones I use Sipsak to display agent login info and their associated queue(s) so that it's easy for agents to know what their status is. -Brodie On Thursday 24 August 2006 10:33 am, John D. Coleman wrote: I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue. The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the queue. I tried using ChanSpy, but then SendText will send messages only to and from the caller who initiated the ChanSpy. One way I could get around this is if I found out how to use SendText from the commandline, like smsq. I'm pretty sure that's not possible because of the nature of SIP MESSAGE but I figured I'd ask. Thanks, John Coleman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendtext() to another machine
I am using sendtext() across 2 machines that connect with IAX. I am sending a text message to the phone. My problem is that I do not answer the text page but after ringing 5 times the IAX connection says it was answered. When I generate this page local to server it works correctly. It does not say answered until I answer it. This only happens over the IAX2 connect between machines. Can I not do sendtext() with IAX across asterisk boxes? -- log starts here. -- Attempting call on IAX2/boxa_to_server/593 for [EMAIL PROTECTED]:1 (Retry 1) -- Call accepted by XX.XX.XX.XX (format alaw) -- Format for call is alaw Channel IAX2/boxa_to_server-1 was answered. -- Executing SendText(IAX2/boxa_to_server-1, 4 t1) in new stack -- Executing AGI(IAX2/boxa_to_server-1, smvoice|-digium_success) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice -- AGI Script smvoice completed, returning 0 -- Executing Hangup(IAX2/boxa_to_server-1, ) in new stack == Spawn extension (smvoice-dialout, smvoice_sendtext, 3) exited non-zero on 'IAX2/boxa_to_server-1' -- Hungup 'IAX2/boxa_to_server-1' - thanks jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages onaSIPhandset's screen
Actually that worked perfectly, now I have another issue. I don't know the parking system too well. I'm not sure whether I should hack res_features.c to include a ast_sendtext() call to peer to send the message or if I can do it from the conf file through SendText(). the issue is whether the conf file knows who parked the call in order to send them the message or whther this is something that happens in res_features.c On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Guillermo Roditi[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List - Non-CommercialDiscussion [mailto: asterisk-users@lists.digium.com]Sent: Fri, 28 Jul 200617:51:26 -0300Subject: Re: [asterisk-users] SendText() displaying textmessages on aSIPhandset's screen for amessage that says test test -- MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.2.13:32827 ;branch=z9hG4bK.39f5be5f;rport;alias To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: sipsak 0.9.6 From: sip:[EMAIL PROTECTED]:32827;tag=1945b6c2 Content-Length: 9 Content-Disposition: desktop test test Ah, it must be the:Content-Disposition: desktopThat does it... interesting. You may be able to hack chan_sip up a bit and add that header in.Joshua ColpDigium___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendtext or sip message - where in RFC
I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Can someone point me where that information is for a SIP message? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendtext or sip message - where in RFC
I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Because sendtext.c is not SIP specific you will not find a reference to SIP related information there. chan_sip.c has a reference to RFC 3428 (http://www.rfc-editor.org/rfc/rfc3428.txt). Have a look at the comment of the function receive_message() in chan_sip.c. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText() displaying text messages on a SIP handset's screen
Hi, I moved a thread over to this list earlier today and since then I have been toying with SendText() which, unfortunately sends what i can only guess is a landline SMS to my SIP handset. I was hoping there would be a way to display custom message on the handset's display. is anyone aware of how this could be done? I was previously trying to use sipsak through execl in res_features.c but that didnt seem very fruitful. also if anyone who knows where i can find some good thoroough documentation on what exactly park and announce is and how to enable it i'd really appreciate it. I am new to asterisk and i am finding myself pretty lost.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages on a SIPhandset's screen
- Original Message - From: Guillermo Roditi [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Fri, 28 Jul 2006 14:26:48 -0300 Subject: [asterisk-users] SendText() displaying text messages on a SIP handset's screen Hi, I moved a thread over to this list earlier today and since then I have been toying with SendText() which, unfortunately sends what i can only guess is a landline SMS to my SIP handset. SendText is generic, all it does is call the function responsible for sending text of the channel (ie: SIP). Now, the SIP channel driver does have the capability to send text messages but what the phone does with them is up to it. Did you try a sip debug to see if chan_sip was indeed sending your phone a message? I was hoping there would be a way to display custom message on the handset's display. is anyone aware of how this could be done? I was previously trying to use sipsak through execl in res_features.c but that didnt seem very fruitful. also if anyone who knows where i can find some good thoroough documentation on what exactly park and announce is and how to enable it i'd really appreciate it. I am new to asterisk and i am finding myself pretty lost.. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages on a SIPhandset's screen
The phone does get the message. Immediately after SendText() happens in the conf the little message light lights up and if I click it then I get the message. I was however looking for a way to write to thephone's display as if it were a console. The only way I've found so far has been with sipsak, which wasn't really working out too well for me so I was hoping I could do this from inside asterisk rather than execl()ing programs from whithin asterisk. On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Guillermo Roditi[mailto:[EMAIL PROTECTED]]To: asterisk-users@lists.digium.comSent: Fri, 28 Jul 2006 14:26:48 -0300Subject: [asterisk-users] SendText() displayingtext messages on a SIPhandset's screen Hi, I moved a thread over to this list earlier today and since then I have been toying with SendText() which, unfortunately sends what i can only guess is a landline SMS to my SIP handset.SendText is generic, all it does is call the function responsible for sending text of the channel (ie: SIP). Now, the SIP channel driver does have the capability to send text messages but what the phone does with them is up to it. Did you try a sip debug to see if chan_sip was indeed sending your phone a message? I was hoping there would be a way to display custom message on the handset's display. is anyone aware of how this could be done? I was previously trying to use sipsak through execl in res_features.c but that didnt seem very fruitful. also if anyone who knows where i can find some good thoroough documentation on what exactly park and announce is and how to enable it i'd really appreciate it.I am new to asterisk and i am finding myself pretty lost.. Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages on aSIPhandset's screen
- Original Message - From: Guillermo Roditi [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 15:09:13 -0300 Subject: Re: [asterisk-users] SendText() displaying text messages on a SIPhandset's screen The phone does get the message. Immediately after SendText() happens in the conf the little message light lights up and if I click it then I get the message. I was however looking for a way to write to thephone's display as if it were a console. The only way I've found so far has been with sipsak, which wasn't really working out too well for me so I was hoping I could do this from inside asterisk rather than execl()ing programs from whithin asterisk. What does the packet look like that you are sending via sipsak? Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages on aSIPhandset's screen
for amessage that says test test--MESSAGE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 10.0.2.13:32827 ;branch=z9hG4bK.39f5be5f;rport;aliasTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 1 MESSAGEContent-Type: text/plain Max-Forwards: 70User-Agent: sipsak 0.9.6From: sip:[EMAIL PROTECTED]:32827;tag=1945b6c2Content-Length: 9Content-Disposition: desktoptest testOn 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Guillermo Roditi[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List - Non-CommercialDiscussion [mailto: asterisk-users@lists.digium.com]Sent: Fri, 28 Jul 200615:09:13 -0300Subject: Re: [asterisk-users] SendText() displaying textmessages on a SIPhandset's screen The phone does get the message. Immediately after SendText() happens in the conf the little message light lights up and if I click it then I get the message. I was however looking for a way to write to thephone's display as if it were a console. The only way I've found so far has been with sipsak, which wasn't really working out too well for me so I was hoping I could do this from inside asterisk rather than execl()ing programs from whithin asterisk.What does the packet look like that you are sending via sipsak? Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText() displaying text messages onaSIPhandset's screen
- Original Message - From: Guillermo Roditi [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 17:51:26 -0300 Subject: Re: [asterisk-users] SendText() displaying text messages on aSIPhandset's screen for amessage that says test test -- MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.2.13:32827;branch=z9hG4bK.39f5be5f;rport;alias To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: sipsak 0.9.6 From: sip:[EMAIL PROTECTED]:32827;tag=1945b6c2 Content-Length: 9 Content-Disposition: desktop test test Ah, it must be the: Content-Disposition: desktop That does it... interesting. You may be able to hack chan_sip up a bit and add that header in. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SendText
Hello, i dont get this feature, how can i send a text to a certain SIP-phone that support this kind of messaging. The WIKI shows an example, but it shows how the receiving phone got to make a call to receive a message. Thx for a hint! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SendText
I used the outgoing spool directory, added a variable like TEXT=Hello world and going to context send_my_text. tehn send_my_text has exten = s,1,SendText($TEXT) Works great. Jerry --- Hello, i dont get this feature, how can i send a text to a certain SIP-phone that support this kind of messaging. The WIKI shows an example, but it shows how the receiving phone got to make a call to receive a message. Thx for a hint! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SendText
Hmm, thx for answering... i understand the idea with the spool directory..When somebody wants to send a message, the variable in spool directorry gets set to message-text. But how can i use the new context, e.g. i want to sent Hello to sip:[EMAIL PROTECTED] ? Christian Jerry Geis schrieb: I used the outgoing spool directory, added a variable like TEXT=Hello world and going to context send_my_text. tehn send_my_text has exten = s,1,SendText($TEXT) Works great. Jerry --- Hello, i dont get this feature, how can i send a text to a certain SIP-phone that support this kind of messaging. The WIKI shows an example, but it shows how the receiving phone got to make a call to receive a message. Thx for a hint! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SendText
Ok, i did some research and its working fine - is there a way to change the change the callerID to something like: MessageCenter or something like this? I always get this realm asterisk. is it the realm, right ? Christian Jerry Geis schrieb: I used the outgoing spool directory, added a variable like TEXT=Hello world and going to context send_my_text. tehn send_my_text has exten = s,1,SendText($TEXT) Works great. Jerry --- Hello, i dont get this feature, how can i send a text to a certain SIP-phone that support this kind of messaging. The WIKI shows an example, but it shows how the receiving phone got to make a call to receive a message. Thx for a hint! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SendText
On 18:39, Fri 24 Jun 05, Christian Hiller wrote: Ok, i did some research and its working fine - is there a way to change the change the callerID to something like: MessageCenter or something like this? I always get this realm asterisk. is it the realm, right ? Christian If it's a callfile you can use the keyword callerid Else do it in the context in extensions.conf: s,1,SetCallerID(your name) s,2,SendText($TEXT) Jerry Geis schrieb: I used the outgoing spool directory, added a variable like TEXT=Hello world and going to context send_my_text. tehn send_my_text has exten = s,1,SendText($TEXT) Works great. Jerry --- Hello, i dont get this feature, how can i send a text to a certain SIP-phone that support this kind of messaging. The WIKI shows an example, but it shows how the receiving phone got to make a call to receive a message. Thx for a hint! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sendtext to a phone that is off
I am doing a sendtext() message to a phone. It works just fine. However I am looking at the case where the phone is off or not available... In that off case voicemail picks up for the phone so the call is still answered and sendtext() is still called. How do I tell if I am getting voicemail so I continue to try and send the text message until it is delivered... vs.. a false sendtext() message that really wasnt delivered at all because voicemail answered the call.. THanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SendText application
Hi All How do I use the sendtext app. In asterisk, what is the syntax? I would like to send a text message to a SIP phone when a specific extension is dialed. Thanks KF ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SendText
On Sun, 2004-11-14 at 13:02, Alessandro Gatti wrote: Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex Well, like most applications it performs on the channel that called it. That means the caller in the terms you used. So when you dial extension 123 in your example the SendText() application will send hello world to you since you are the channel that executed it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SendText
That makes sense. I will need to figure out how to use it send it to the callee.. Thanks, Alessandro -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Monday, November 15, 2004 7:51 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SendText On Sun, 2004-11-14 at 13:02, Alessandro Gatti wrote: Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex Well, like most applications it performs on the channel that called it. That means the caller in the terms you used. So when you dial extension 123 in your example the SendText() application will send hello world to you since you are the channel that executed it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SendText
Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
