Re: [asterisk-users] Respond to an out of call SIP MESSAGE
Yes, that is how the problem was solved before when the transport protocol for messages was UDP, but TLS (TCP) this doesn't work. Partly because MessageSend doesn't support SIPS and partly because it focuses on sending a new message instead of reusing an existing stream. Using wireshark I can see that the TLS stream is still alive as there is no tear down communication. The "same,n," is a good shortcut to make the text more readable, I'll add that but I don't think that is the cause as I can follow the source code and see that the SendText command is executed but fails silently. Thanks for the feedback, Emil From: Matthew Jordan Sent: Monday, September 21, 2015 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Emil Ohlsson Subject: Re: [asterisk-users] Respond to an out of call SIP MESSAGE On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain wrote: > On Mon, 21 Sep 2015 06:48:52 + > Emil Ohlsson 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 an expert but perhaps you want this. > > [sip-im] > exten s,1,NoOp(Got message) > same,n,Answer() > same,n,Agi(agi://localhost/messagehandler.agi?...) > same,n,SendText(Message received) > > Replacing "exten _X!" with "same" is just a shortcut. I find that > there are lots of places where spaces cause problems so I just remove > them all for good measure. Finally, I am not sure what the mechanism > is here but if it is like a goto then I think that you want the 's' > priority. > > Or, I totally don't know what I am talking about and my education will > be advanced by the replies to this message. :-) > If you want to send an out of call SIP MESSAGE request, you'll need to use the MessageSend application: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend SendText is used for sending text messages within a call. Since a SIP channel is not servicing the out of call text message, you cannot use it to send a SIP MESSAGE request back to whatever sent the original SIP MESSAGE request. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://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] How to config instance messaging for asterisk 12
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 default extension.conf from the installation and I successfully could make the call over each but when I try to send message, it does not receive by the client. -- _ -- 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] Brazil TDM routes
Dear fellows, how are you? I´m offering TDM routes for Brazil (landline and mobile destinations) with low prices, TDM ccts (no GSM), ASR and ACD great. Pre paid, by paypal. If you have interest, please just let me know. With Best Regards Josue -- _ -- 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] Respond to an out of call SIP MESSAGE
On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain wrote: > On Mon, 21 Sep 2015 06:48:52 + > Emil Ohlsson 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 an expert but perhaps you want this. > > [sip-im] > exten s,1,NoOp(Got message) > same,n,Answer() > same,n,Agi(agi://localhost/messagehandler.agi?...) > same,n,SendText(Message received) > > Replacing "exten _X!" with "same" is just a shortcut. I find that > there are lots of places where spaces cause problems so I just remove > them all for good measure. Finally, I am not sure what the mechanism > is here but if it is like a goto then I think that you want the 's' > priority. > > Or, I totally don't know what I am talking about and my education will > be advanced by the replies to this message. :-) > If you want to send an out of call SIP MESSAGE request, you'll need to use the MessageSend application: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend SendText is used for sending text messages within a call. Since a SIP channel is not servicing the out of call text message, you cannot use it to send a SIP MESSAGE request back to whatever sent the original SIP MESSAGE request. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://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] Call waiting for Queue Agents.
Hi, Thanks for your reply. It's working. I forgot to enable call waiting under extensions in Asterisk. Best regards On Mon, Sep 21, 2015 at 3:36 PM, Ishfaq Malik wrote: > > > On 21 September 2015 at 15:27, Aziz TestAccount > wrote: > >> Hi All, >> >> I have a question about the Queues. >> >> I'm using Asterisk 11.13.0 , and I want to configure the following setup >> : >> >> When there is an incoming call to the queue all agents should ring even >> those that are already in call, they should receive a second call. >> >> Is this doable in any Asterisk version ? >> >> Thanks in advance. >> >> >> > In 1.8 there is a ring in use option at the queue level. I doubt this will > have been removed in 11. > > ; If you want the queue to avoid sending calls to members whose devices are > ; known to be 'in use' (via the channel driver supporting that device > state) > ; uncomment this option. (Note: only the SIP channel driver currently is > able > ; to report 'in use'.) > ; > ; ringinuse = no > > > Regards > > Ish > > > -- > > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)161 660 2350 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, Duplex 2, Ducie House > 37 Ducie Street > Manchester, M1 2JW > COMPANY REG NO. 04920552 > > > -- > _ > -- 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 > -- _ -- 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] Respond to an out of call SIP MESSAGE
On Mon, 21 Sep 2015 06:48:52 + Emil Ohlsson 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 an expert but perhaps you want this. [sip-im] exten s,1,NoOp(Got message) same,n,Answer() same,n,Agi(agi://localhost/messagehandler.agi?...) same,n,SendText(Message received) Replacing "exten _X!" with "same" is just a shortcut. I find that there are lots of places where spaces cause problems so I just remove them all for good measure. Finally, I am not sure what the mechanism is here but if it is like a goto then I think that you want the 's' priority. Or, I totally don't know what I am talking about and my education will be advanced by the replies to this message. :-) -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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] Call waiting for Queue Agents.
On 21 September 2015 at 15:27, Aziz TestAccount wrote: > Hi All, > > I have a question about the Queues. > > I'm using Asterisk 11.13.0 , and I want to configure the following setup : > > When there is an incoming call to the queue all agents should ring even > those that are already in call, they should receive a second call. > > Is this doable in any Asterisk version ? > > Thanks in advance. > > > In 1.8 there is a ring in use option at the queue level. I doubt this will have been removed in 11. ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Call waiting for Queue Agents.
Hi All, I have a question about the Queues. I'm using Asterisk 11.13.0 , and I want to configure the following setup : When there is an incoming call to the queue all agents should ring even those that are already in call, they should receive a second call. Is this doable in any Asterisk version ? Thanks in advance. -- _ -- 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] Realtime Voicemail MWI
I'm bone stock except for the following global(s). rtcachefriends=yes limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes notifybusy=yes My Sip Table looks like (Sorry, Hard to read, Note the 207@103 mailbox setting. 90103_14"Nick" <321XXX>103_14 SECRETPASS 103_internalnorfc2833dynamic port,invite207@103force_rport,comedia yes9001300friendallulaw0yes0 2070 And my voicemail table (Hard to read as well) 98207103207PASSNickcentralnoyes nononono1nonoyesnonoyes 2015-08-17 15:22:09yesno4 25 Forcing a reload from the DB with the Prune>Load method loads the mailbox about 20% of the time Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 1/0 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "Nick" <321XXX> Nick Olsen Network Operations (855) FLSPEED x106 From: "Stefan Tichy" Sent: Sunday, September 20, 2015 9:28 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime Voicemail MWI On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote: > I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, > These are loaded into asterisk without the mailbox info. Leading to > "Received SIP subscribe for peer without mailbox" notices. And non-working > MWI. > > Occasionally, It just works. But only on a peer or two at a time. And > it'll stop working after a few minutes. Here it seems to be the other way round. Occasionally I see that peers have lost there mailbox setting and don't get notify messages with voicemail information. It is Asterisk 13.5.0 "sip prune realtime peer ..." "sip show peer ... load" After this the setting is restored, but until now I have no idea why this happens. The database field mailbox remains unchanged. Could you post the Realtime SIP Settings? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- 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 -- _ -- 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] Dahdi modules error with OpenVZ kernel on Debian 7 (2.6.32-openvz-042stab111.11-amd64)
Hi! After upgrading kernel from 2.6.32-openvz-042stab108.8-amd64 to 2.6.32-openvz-042stab111.11-amd64 I fail to load the DAHDI kernel modules. It's the package dahdi-linux-complete-2.10.2+2.10.2 and it looks like it compiles and installs just fine (a few error in the text indicates that something is wrong however) After "make ; make install ; modprobe dahdi" This is the output: libkmod: ERROR ../libkmod/libkmod.c:505 kmod_lookup_alias_from_builtin_file: could not open builtin file '/lib/modules/2.6.32-openvz-042stab111.11-amd64/modules.builtin.bin' FATAL: Module dahdi not found. When I look in /lib/modules/2.6.32-openvz-042stab111.11-amd64 the dahdi-subtree that exists in the /lib/modules/2.6.32-openvz-042stab108.8-amd64 (previous kernel) is missing. Output for make is attached below. Anyone that is a little bit less lost with compiling kernel modules than I am? Many greetings! /Johan --- uname -a Linux pi 2.6.32-openvz-042stab111.11-amd64 #1 SMP Tue Sep 1 18:27:11 MSK 2015 x86_64 GNU/Linux root@pi:/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' VERSION /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/xpp/xpp_version.h Building modules, stage 2. MODPOST 36 modules WARNING: could not find /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' root@pi:/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux# make install make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' VERSION /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/xpp/xpp_version.h Building modules, stage 2. MODPOST 36 modules WARNING: could not find /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' build_tools/uninstall-modules dahdi 2.6.32-openvz-042stab111.11-amd64 make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes INSTALL_MOD_PATH= INSTALL_MOD_DIR=dahdi modules_install expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dummy.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_eth.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_ethmf.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_loc.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_jpah.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_kb1.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_mg2.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_sec.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_sec2.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_transcode.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_vpmadt032_loader.ko INSTALL /u
Re: [asterisk-users] Respond to an out of call SIP MESSAGE
Could it be in the [general] section you should have; accept_outofcall_message=yes Your line appears to be missing the 'p' in accept and an extraneous 's' in message. Larry. On 21/09/2015 2:48 PM, Emil Ohlsson wrote: 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 [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 can see in the log from Asterisk that the script in the sip-im context is running, but there is no message sent. I have followed the code in the call, and it seems like there is no channel registered with the SendText application. Is there some other approach that I could use to send a SIP MESSAGE back to the client? Does the client need to register for this to work? BR, Emil -- _ -- 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
