similar to: Commas is variables problem

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 =>

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

Hello I have a simple 1 channel goip gateway (http://www.voip-info.org/wiki/view/GoIP). The incoming and outgoing calls work with Asterisk except the caller ID for the outgoing calls. I think I have exhausted all possible options regarding setting a caller ID and it still doesn't work. The recipients will get "private number". The incomings caller ids are work just fine. exten

I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does

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 all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks

I have tried both by hand and hitting tab to auto complete: *CLI> channel request hangup Message/ast_msg_queue Message/ast_msg_queue is not a known channel On 31/10/19 14:18, Sean Bright wrote: > On 10/31/2019 2:13 PM, Carlos Chavez wrote: >> I assume this is something created by Freepbx.  If I do a "channel >> request hangup" it tells me the channel does not exist.

On Thu, Feb 6, 2020 at 12:34 PM sduthil at wazo.io <sduthil at wazo.io> wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > For those of you who actually process SIP MESSAGE requests... Do you > > use any of the AMI events generated by the "Message/ast_msg_queue" > > channel? We want to change that channel to an "internal" channel that > >

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/stefan at my.asterix.pbx and it would go like this: [incoming_guest]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

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

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi folks, some of you might know Sergio Chersovani's rewrite of chan-sccp, the asterisk channel driver for Cisco Skinny phones. I have put up an unofficial site with some sample configs, a little help and a webbased forum. Both are just new, so don't expect too much :-). Everybody is invited to participate especially at the forum. Any comments, proposals, critics are very welcome. Find

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have no audio on chan_mobile but this message repeats continuously: Dropping incompatible voice frame on Mobile/DNA-54f4 of format slin since our native format has changed to 0x0 (nothing) Can somebody point me to the right direction? Asterisk SVN-branch-1.6.2-r313579 - -Stefan - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\

Hi, I'm playing with * for about 12 years now and since about 10 years, it's my home PBX. I can do pretty much everything I want but one thing I haven't managed yet... Mobile connection via bluetooth... I'm still using a Nokia E51 and the setup and everything works fine. However, on the second or third call, the incoming audio is noise. I have tried alignmentdetection=yes and also

Hi, I'm tinkering with Asterisk for * for about 12 years now and since about 10 years, it's my home PBX. I was off the list for something like 7 years - had other things to do. But... I remember, then, sometimes came over 1000 mails in 24h. Now it's hardly 50 new mails per week. Is the list dead? Or is the project dead? Or is nobody tinkering any more and everybody buying some

On 01/08/2015 11:37 PM, ricky gutierrez wrote: > Hi list , someone on the list has seen this type of connection > attempts in asterisk, fail2ban does not stop > > 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: >

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,

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