similar to: Asterisk compatibility with SMS services

> Can anyone comment on using SMS in conjunction with VoIP service using > one of these three VoIP providers: voip.ms, vitelity.com, > flowroute.com? Are some SMS services more compatible with Asterisk > (i.e. SMS over SIP works perfectly or not)? Is it best to use a > different data channel for SMS messages (i.e. SMS via HTTP, SMS via > XMPP) instead of Asterisk's built

Hello My provider allows to activate/deactivate a forwarding rule by sending a SIP MESSAGE. This is done outside a call. That is, while there is no ongoing call, a SIP client just sends the following message: MESSAGE sip:543951354657 at callfwd.sip.providerx.com SIP/2.0 Call-ID: b9ba106e-613a-46b9-8a4d-0efb4dc0a0f2 CSeq: 1 MESSAGE To: <sip:543951354657 at

Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText" Channel: SIP/600" Body: This is a test. Message: This is a test. Extension: 600";

On 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends

Yes, sorry actually in asterisk 13, anyway how could i do that ? On Tue, Sep 22, 2015 at 5:43 PM, Joshua Colp <jcolp at digium.com> wrote: > On 15-09-22 03:34 AM, Thyda ENG wrote: > >> I am using the asterisk 12 with pjsip, I wonder how could I config the >> instance meesseging for pjsqip in asterisk 12 ? What is the default >> message context for pjssip ? I use the

MessageSend is command for send message, however I don't know what the context for sending message. I create a pjsip with the context 'from-internal' then when i config the extension for context 'from-internal' it works but then the my call dialplan does not work. Because they both sms and call are coming to the same context 'from-internal', as I notice. I wonder how

I am using the asterisk 13 and I config my dialplan for the SIP messaging as the following : http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html [astsms] exten => _.,1,NoOp(SMS receiving dialplan invoked) exten => _.,n,NoOp(To ${MESSAGE(to)}) exten => _.,n,NoOp(From ${MESSAGE(from)}) exten => _.,n,NoOp(Body ${MESSAGE(body)}) exten =>

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

Thanks again. How do you create that message context in extensions.conf? On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda at gmail.com> wrote: > According to what I have done , I add the message_context to the > pjsip.endpoint_custom.conf in /etc/asterisk and then I create that > message_context in the extension.conf, and it works. > > On Tue, Nov 17, 2015 at 9:34 AM,

Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603 at DLPN_AlDimnaDialPlan:601] Dial("SIP/601-00000002", "SIP/603") in new stack [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433 dial_exec_full: Unable to

Hi. I have read in some web sites that ASTERISK can support messages out of calls. What does it exactly means? 1 - Can a dialplan script accept and handle a message from a callee party, even before the call be connected? 2 - Can a ringing callee send SIP MESSAGE to the ASTERISK even before answer the call? 3- Could I use dialplan function MESSAGE() to receive SIP messages from callees, even

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same => n,SMS(hello,a,17654307001,"hello nick") - nick

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

Basically I've found that the 2.0.3 version of Samba is not supporting shadowed password files on SCO Unix 3.2v4.2. It appears to be doing it on SCO Unix OSR5 however. This shows up both in smbd and swat. We found this out by playing with the newest version on a SCO Unix OSR5 box. I recompiled the software on that box for a SCO 3.2v4.2 box but was unable to get the resulting binary to work

Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after