[asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation ( http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. Here's my sip.conf: bindport = 5070 ;Kamailio is at port 5060, and it's always used as outbound proxy bindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407744248 defaultuser: 660 fullcontact: sip:6...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL avpf: yes force_avp: yes icesupport: yes directmedia: yes encryption: yes nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL setvar: NULL callerid: NULL amaflags: NULL videosupport: no maxcallbitrate: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: md5ofmypwd rpid: NULL domain: testers.com sippasswd2: NULL This is how all other clients are currently defined: id: 7 name: 771 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407748788 defaultuser: 771 fullcontact: sip:7...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL avpf: no force_avp: NULL icesupport: NULL directmedia: yes encryption: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL setvar: NULL callerid: NULL amaflags: NULL videosupport: NULL maxcallbitrate: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: NULL dtlsenable: NULL dtlsverify: NULL dtlsprivatekey: NULL dtlssetup: NULL dtlscertfile: NULL dtlscafile: NULL sippasswd: 27e13af7c596313350986c58c9d24946 rpid: NULL domain: testers.com sippasswd2: NULL cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar
Re: [asterisk-users] Dahdi CAPI migration
Hello The answers to your questions are: 1, OS CentOS release 5.5 (Final) Trixbox installed at: Autogenerated by /usr/sbin/dahdi_genconf on Fri Nov 25 18:03:26 2011 2, Kernel Linux 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:53:10 EST 2010 i686 i686 i386 GNU/Linux 3, Packages asterisk16-dahdi.i3861.6.0.26-1_trixboxinstalled dahdi-firmware.noarch2.0.0-1_centos5 installed dahdi-firmware-oct6114-064.noarch1.05.01-1_centos5 installed dahdi-firmware-oct6114-128.noarch1.05.01-1_centos5 installed dahdi-firmware-tc400m.noarch MR6.12-1_centos5 installed dahdi-firmware-vpmadt032.noarch 1.07-1_centos5installed dahdi-linux.i386 2.3.0.1-1_trixbox installed dahdi-tools.i386 2.3.0-1_trixbox installed dahdi-tools-doc.i386 2.2.0-4_trixbox installed kmod-dahdi-linux.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 kmod-dahdi-linux-xen.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 dahdi-linux-devel.i386 2.3.0.1-1_trixbox trixbox28 kmod-dahdi-linux-PAE.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 libpri.i386 1.4.10.2-1_centos5installed libpri-devel.i3861.4.10.2-1_centos5trixbox28 asterisk16.i386 1.6.0.26-1_trixboxinstalled kmod-mISDN.i686 1.1.7.2-4_centos5.2.6.18_164.11.1.el5 kmod-mISDN-xen.i686 1.1.7.2-3_centos5.2.6.18_164.11.1.el5 mISDN.i386 1.1.7.2-4_centos5 installed mISDNuser.i386 1.1.7.2-2_centos5 installed asterisk-chan_misdn.i386 1.4.22-3 trixbox kmod-mISDN-PAE.i686 1.1.7.2-3_centos5.2.6.18_164.11.1.el5 mISDN.i686 1.1.7-27 trixbox mISDN-debuginfo.i686 1.1.7-24 trixboxaddons mISDN-devel.i686 1.1.7-27 trixbox mISDN-devel.i386 1.1.7.2-4_centos5 trixbox28 mISDN-kmod-base.i686 1.1.7.2-1_centos5.2.6.18_128.1.10.el5 mISDN-modules.i686 1.1.7-27.2.6.18_92.1.18.el5 trixbox mISDNuser-debuginfo.i386 1.1.7-15 trixboxaddons mISDNuser-devel.i386 1.1.7.2-2_centos5 trixbox28 Asterisk 1.6.0.26-FONCORE-r78, Copyright (C) 1999 - 2010 Digium, Inc. and others. 4, What do you mean with the OS-es were clones ...? Did you create an image of the old Trixbox machine and installed that on the new machine? It means that they are Xen virtual machines, exact bit by bit vm clones so they should have all the same configuration files, run the exact same Xen kernels. What complicates things a bit, and probably the cause of my errors is Xen's PCI passthrough. The only reason why I use something so obsolete like Xen is just this feature otherwise I would be using kvm, vmware, virtualbox or whatever virt technologies but for those you must have vt(d) hardware support and the machine I dealing with here doesn't have this, neither the old one. 5, Lsdadhi (this is on the first, working machine) ### Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS 1 BRIClear (In use) (SWEC: MG2) 2 BRIClear (In use) (SWEC: MG2) 3 BRIHardware-assisted HDLC (In use) ### Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS 4 BRIClear (In use) (SWEC: MG2) 5 BRIClear (In use) (SWEC: MG2) 6 BRIHardware-assisted HDLC (In use) ### Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS RED 7 BRIClear (In use) (SWEC: MG2) RED 8 BRIClear (In use) (SWEC: MG2) RED 9 BRIHardware-assisted HDLC (In use) RED ### Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS 10 BRIClear (In use) (SWEC: MG2) 11 BRIClear (In use) (SWEC: MG2) 12 BRIHardware-assisted HDLC (In use) 6, Asterisk logs (new machine when it failed) full.4:[Aug 7 12:39:58] WARNING[1654] chan_dahdi.c: Unable to specify channel 1: No such device or address full.4:[Aug 7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to open channel 1: No such device or address full.4:[Aug 7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to register channel '1-2' full.4:[Aug 7 12:39:58] VERBOSE[1654] logger.c: -- codec_dahdi: using generic PLC full.4:[Aug 7 12:39:58] ERROR[1654] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory full.4:[Aug 7 12:39:58] VERBOSE[1654] logger.c: codec_dahdi.so =
[asterisk-users] MeetMe - Howto put in talk only mode using CLI/AMI
Hi, is there a way to put a conference participant in talk only mode (not listening) using CLI or AMI like mute/unmute ? MeetMe in Asterisk 1.8 Thanks for any hint. -- Daniel -- _ -- 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] Sending and receiving fax with Digium FFA
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine with
Asterisk 1.8. I have managed to register and installed the Digium
modules. Sending and receiving through it have resulted in failure. The
output of fax show capabilities is:
Registered FAX Technology Modules:
Type: DIGIUM
Description : Digium FAX Driver
Capabilities: SEND RECEIVE T.38 G.711 MULTI-DOC
1 registered modules
We have a fax blackbox through which I'm trying to send faxes to the
Asterisk server. Every time that I send a fax I get a timeout error.
Been tinkering with the settings and whatnot to get it working.
The extension to receive fax:
exten = recvfax,1,Verbose(2,Receiving fax)
same = n,Set(FAXDEST=/tmp/fax)
same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
same = n,Wait(8)
same = n,ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d)
It's without most of the tinkering I've done, which are: setting ecm to
no, tweaking the min/max rate and other things.
Also, because the fax machine can't print (half broken), we receive our
faxes through a fax to email service we have subscribed to, so the tests
for sending have that one as a destination.
The extension to send fax:
exten = sendfax,1,Verbose(2,Sending fax)
same = n,Set(faxlocation=/tmp)
same = n,Set(faxfile=fax.tiff)
same = n,Set(FAXOPT(headerinfo)=Testing FAX)
same = n,Set(FAXOPT(localstationid)=123456)
same = n,SendFax(${faxlocation}/${faxfile},d)
same = n,Verbose(2, Fax Status: ${FAXOPT(error)})
I did the exact same thing, and tried sending from both a SIP channel
and a DAHDI line. The weird thing is that when I am sending through
Asterisk I get, as a response to fax, a recorded message from the telco.
Sending through the same line with the fax machine works perfectly.
Any advice and help is welcome.
--
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[asterisk-users] NotifyCID to see who is calling for call pickup
Hello, If the phone of my colleague rings, I can see this with BLF-lamps on my Snom IP-phone. I would also like to see *_who_* is calling. I would like to see the external number on my screen so I can choose whether to pickup the call with BLF. Therefore I have in sip.conf : notifycid = yes With this setting on, I see on my screen : 10 -- 10 10 is the internal extension of my colleague. But how can I see the external number that is calling in ? I would expect to see : 10 -- 3221234567 3221234567 being the external number I would like to know about. This is what Asterisk sends to my Snom IP-phone : [Aug 11 16:37:56] Reliably Transmitting (NAT) to my.pub.lic.ip:1024: NOTIFY sip:testacc77003@192.168.1.109:1024 SIP/2.0 Via: SIP/2.0/UDP ip.ast.ser.ver:5060;branch=z9hG4bK5b999cd4;rport Max-Forwards: 70 From: sip:1...@ip.ast.ser.ver;user=phone;tag=as6b302bda To: sip:testacc77...@ip.ast.ser.ver;tag=ashydm1he5 Contact: sip:1...@ip.ast.ser.ver:5060 Call-ID: 3c26b7878939-ri1v0tkqfa2h CSeq: 103 NOTIFY User-Agent: pbx Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 516 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=1 state=full entity=sip:1...@ip.ast.ser.ver dialog id=10 call-id=pickup-3c26b7878939-ri1v0tkqfa2h local-tag=ashydm1he5 remote-tag=as6b302bda direction=recipient remote identity display=10sip:1...@ip.ast.ser.ver/identity target uri=sip:1...@ip.ast.ser.ver/ /remote local identitysip:1...@ip.ast.ser.ver/identity target uri=sip:1...@ip.ast.ser.ver/ /local stateearly/state /dialog /dialog-info Where does Asterisk take the information to put into the dialog-info and how can I change it to display the external number also ? Thanks ! Jonas. -- _ -- 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] Sending and receiving fax with Digium FFA
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine
with Asterisk 1.8. I have managed to register and installed the
Digium modules. Sending and receiving through it have resulted in
failure. The output of fax show capabilities is:
Registered FAX Technology Modules:
Type: DIGIUM
Description : Digium FAX Driver
Capabilities: SEND RECEIVE T.38 G.711 MULTI-DOC
1 registered modules
We have a fax blackbox through which I'm trying to send faxes to
the Asterisk server. Every time that I send a fax I get a timeout
error. Been tinkering with the settings and whatnot to get it working.
The extension to receive fax:
exten = recvfax,1,Verbose(2,Receiving fax)
same = n,Set(FAXDEST=/tmp/fax)
same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
same = n,Wait(8)
same = n,ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d)
It's without most of the tinkering I've done, which are: setting ecm
to no, tweaking the min/max rate and other things.
Also, because the fax machine can't print (half broken), we receive
our faxes through a fax to email service we have subscribed to, so
the tests for sending have that one as a destination.
The extension to send fax:
exten = sendfax,1,Verbose(2,Sending fax)
same = n,Set(faxlocation=/tmp)
same = n,Set(faxfile=fax.tiff)
same = n,Set(FAXOPT(headerinfo)=Testing FAX)
same = n,Set(FAXOPT(localstationid)=123456)
same = n,SendFax(${faxlocation}/${faxfile},d)
same = n,Verbose(2, Fax Status: ${FAXOPT(error)})
I did the exact same thing, and tried sending from both a SIP
channel and a DAHDI line. The weird thing is that when I am sending
through Asterisk I get, as a response to fax, a recorded message
from the telco. Sending through the same line with the fax machine
works perfectly.
Any advice and help is welcome.
Can you post the output of the Asterisk CLI from a failed fax call? What
you have looks ok for the most part, at least on the receiving end. Did
you install the license key for the Free FAX for Asterisk module?--
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[asterisk-users] Asterisk support for Bittorrent Bleep
Hello, Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent Bleep (a private P2P SIP-based messaging application in early alpha) http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/ I have personally been a fan of Asterisk and have been using it for years and now that we have (kind of) released Bleep, I wanted to ask you guys to let us know what you think. Considering that Bleep is built on an engine (think of it as a distributed SIP proxy) that supports SIP, I thought it might be beneficial to ask you guys for your ideas. Here is what I have in mind but will be happy to hear your thoughts on everything that is relevant to Bleep and Asterisk: 1- What do you think about supporting Bleep in Asterisk? Similar to Skype channels but way more flexible (considering the interface will be SIP). Our engine can take care of all lookups, NAT traversals, encryption, etc. We can essentially enable Asterisk connected devices to be able to talk to Bleep users. 2- How could the Asterisk community benefit from Bleep (or the engine behind it)? 3- what features would you like to see implemented in Bleep (the consumer app) or its engine? Let's see if we can come up with any interesting idea. Thanks in advance. Farid -- _ -- 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] Asterisk support for Bittorrent Bleep
On 11/08/14 16:46, Farid Fadaie wrote: Hello, Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent Bleep (a private P2P SIP-based messaging application in early alpha) http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/ I have personally been a fan of Asterisk and have been using it for years and now that we have (kind of) released Bleep, I wanted to ask you guys to let us know what you think. Considering that Bleep is built on an engine (think of it as a distributed SIP proxy) that supports SIP, I thought it might be beneficial to ask you guys for your ideas. Here is what I have in mind but will be happy to hear your thoughts on everything that is relevant to Bleep and Asterisk: 1- What do you think about supporting Bleep in Asterisk? Similar to Skype channels but way more flexible (considering the interface will be SIP). Our engine can take care of all lookups, NAT traversals, encryption, etc. We can essentially enable Asterisk connected devices to be able to talk to Bleep users. 2- How could the Asterisk community benefit from Bleep (or the engine behind it)? 3- what features would you like to see implemented in Bleep (the consumer app) or its engine? Let's see if we can come up with any interesting idea. Thanks in advance. Farid Given how many SIP based attacks and probes are going around and that most peoples defence is to use something like fail2ban I think your biggest hurdle in getting people to use it will be persuading them that it is secure and giving them the tools to block malicious use. -- _ -- 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] 401 Unathorized
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT) Using INVITE request as basis request - 41597440-0-320116780@carrierIP Found peer 'phonenumber' for 'phonenumber' from carrierIP:5060 --- Reliably Transmitting (NAT) to carrierIP:5060 --- SIP/2.0 401 Unauthorized Good call --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT) Using INVITE request as basis request - 41604639-0-321360830@carrierIP Found peer 'carrierIP' for 'phonenumber' from carrierIP:5060 --- Transmitting (no NAT) to carrierIP:5060 --- Call proceeds here. - It's all coming from the same carrier IP and the same SIP trunk. All are set via static IP (No registrations). And nat is set to no (Everything is on a public IP). Has anyone else run across anything similar? Thanks David -- [image: Ringfree Communications, Inc] http://ringfree.biz/ David Wessell / President 828-575-0030 x101/ da...@ringfree.biz Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830 PO BOX 1994 Hendersonville, NC 28793 http://ringfree.biz This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- _ -- 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] Asterisk support for Bittorrent Bleep
On Monday 11 Aug 2014, Farid Fadaie wrote: Hello, Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent Bleep (a private P2P SIP-based messaging application in early alpha) http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized- communications/ I have personally been a fan of Asterisk and have been using it for years and now that we have (kind of) released Bleep, I wanted to ask you guys to let us know what you think. Considering that Bleep is built on an engine (think of it as a distributed SIP proxy) that supports SIP, I thought it might be beneficial to ask you guys for your ideas. Here is what I have in mind but will be happy to hear your thoughts on everything that is relevant to Bleep and Asterisk: 1- What do you think about supporting Bleep in Asterisk? Similar to Skype channels but way more flexible (considering the interface will be SIP). Our engine can take care of all lookups, NAT traversals, encryption, etc. We can essentially enable Asterisk connected devices to be able to talk to Bleep users. 2- How could the Asterisk community benefit from Bleep (or the engine behind it)? 3- what features would you like to see implemented in Bleep (the consumer app) or its engine? It all sounds interesting. Where is the GIT repo? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Sending and receiving fax with Digium FFA
Hello;
Just taking a quick glance at it, I think you have a syntax error in
your dial plan. Instead of ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d,
shouldn't it be ReceiveFax(${FAXDEST}/${tempfax}.tiff,fd) with no comma
between the f and d options?
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dumitru
Sent: Monday, August 11, 2014 9:39 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sending and receiving fax with Digium FFA
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine with
Asterisk 1.8. I have managed to register and installed the Digium modules.
Sending and receiving through it have resulted in failure. The output of fax
show capabilities is:
Registered FAX Technology Modules:
Type: DIGIUM
Description : Digium FAX Driver
Capabilities: SEND RECEIVE T.38 G.711 MULTI-DOC
1 registered modules
We have a fax blackbox through which I'm trying to send faxes to the
Asterisk server. Every time that I send a fax I get a timeout error. Been
tinkering with the settings and whatnot to get it working.
The extension to receive fax:
exten = recvfax,1,Verbose(2,Receiving fax)
same = n,Set(FAXDEST=/tmp/fax)
same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
same = n,Wait(8)
same = n,ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d)
It's without most of the tinkering I've done, which are: setting ecm to no,
tweaking the min/max rate and other things.
Also, because the fax machine can't print (half broken), we receive our
faxes through a fax to email service we have subscribed to, so the tests for
sending have that one as a destination.
The extension to send fax:
exten = sendfax,1,Verbose(2,Sending fax)
same = n,Set(faxlocation=/tmp)
same = n,Set(faxfile=fax.tiff)
same = n,Set(FAXOPT(headerinfo)=Testing FAX)
same = n,Set(FAXOPT(localstationid)=123456)
same = n,SendFax(${faxlocation}/${faxfile},d)
same = n,Verbose(2, Fax Status: ${FAXOPT(error)})
I did the exact same thing, and tried sending from both a SIP channel and a
DAHDI line. The weird thing is that when I am sending through Asterisk I
get, as a response to fax, a recorded message from the telco. Sending
through the same line with the fax machine works perfectly.
Any advice and help is welcome.
--
_
-- 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
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Re: [asterisk-users] 401 Unathorized
On 8/11/14, 10:57 AM, David Wessell wrote: I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT) Using INVITE request as basis request - 41597440-0-320116780@carrierIP Found peer 'phonenumber' for 'phonenumber' from carrierIP:5060 --- Reliably Transmitting (NAT) to carrierIP:5060 --- SIP/2.0 401 Unauthorized Good call --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT) Using INVITE request as basis request - 41604639-0-321360830@carrierIP Found peer 'carrierIP' for 'phonenumber' from carrierIP:5060 --- Transmitting (no NAT) to carrierIP:5060 --- Call proceeds here. - It's all coming from the same carrier IP and the same SIP trunk. All are set via static IP (No registrations). And nat is set to no (Everything is on a public IP). According to the above, the good call matches a different peer from the bad call. Look into that and you will find the cause. Has anyone else run across anything similar? Thanks David -- Ringfree Communications, Inc http://ringfree.biz/ David Wessell / President 828-575-0030 x101/ da...@ringfree.biz mailto:da...@ringfree.biz Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830 PO BOX 1994 Hendersonville, NC 28793 http://ringfree.biz http://ringfree.biz/ This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- Technical Support http://www.cellroute.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
[asterisk-users] Asterisk 13.0.0-beta1 Now Available!
The Asterisk Development Team is pleased to announce the first beta release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases All interested users of Asterisk are encouraged to participate in the Asterisk 13 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. All Asterisk users are invited to participate in the #asterisk-bugs channel to help communicate issues found to the Asterisk developers. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list (http://lists.digium.com). Asterisk 13 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 11. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 13, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 A short list of new features includes: * Asterisk security events are now provided via AMI, allowing end users to monitor their Asterisk system in real time for security related issues. * Both AMI and ARI now allow external systems to control the state of a mailbox. Using AMI actions or ARI resources, external systems can programmatically trigger Message Waiting Indicators (MWI) on subscribed phones. This is of particular use to those who want to build their own VoiceMail application using ARI. * ARI now supports the reception/transmission of out of call text messages using any supported channel driver/protocol stack through ARI. Users receive out of call text messages as JSON events over the ARI websocket connection, and can send out of call text messages using HTTP requests. * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers. * The PJSIP stack can now be used as a means of distributing device state or mailbox state via PUBLISH requests to other Asterisk instances. This is analogous to Asterisk's clustering support using XMPP or Corosync; unlike existing clustering mechanisms, using the PJSIP stack to perform the distribution of state does not rely on another daemon or server to perform the work. And much more! More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation A full list of all new features can also be found in the CHANGES file: http://svnview.digium.com/svn/asterisk/branches/13/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta1 Thank you for your continued support of Asterisk! -- _ -- 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] Dahdi CAPI migration
On 11-08-14 11:09, Toney Mareo wrote: Hello The answers to your questions are: 1, OS CentOS release 5.5 (Final) That version is ancient and full of security holes. It is recommended to at least update to CentOS 5.10 + updates. That's assuming there are Trixbox kmod-dahdi-linux* RPMs for CentOS 5.10. Trixbox installed at: Autogenerated by /usr/sbin/dahdi_genconf on Fri Nov 25 18:03:26 2011 Trixbox CE no longer exists and is no longer supported. Why continue using it? 2, Kernel Linux 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:53:10 EST 2010 i686 i686 i386 GNU/Linux 3, Packages asterisk16-dahdi.i3861.6.0.26-1_trixboxinstalled dahdi-firmware.noarch2.0.0-1_centos5 installed dahdi-firmware-oct6114-064.noarch1.05.01-1_centos5 installed dahdi-firmware-oct6114-128.noarch1.05.01-1_centos5 installed dahdi-firmware-tc400m.noarch MR6.12-1_centos5 installed dahdi-firmware-vpmadt032.noarch 1.07-1_centos5installed dahdi-linux.i386 2.3.0.1-1_trixbox installed dahdi-tools.i386 2.3.0-1_trixbox installed dahdi-tools-doc.i386 2.2.0-4_trixbox installed kmod-dahdi-linux.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 kmod-dahdi-linux-xen.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 dahdi-linux-devel.i386 2.3.0.1-1_trixbox trixbox28 kmod-dahdi-linux-PAE.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5 libpri.i386 1.4.10.2-1_centos5installed libpri-devel.i3861.4.10.2-1_centos5trixbox28 asterisk16.i386 1.6.0.26-1_trixboxinstalled kmod-mISDN.i686 1.1.7.2-4_centos5.2.6.18_164.11.1.el5 kmod-mISDN-xen.i686 1.1.7.2-3_centos5.2.6.18_164.11.1.el5 mISDN.i386 1.1.7.2-4_centos5 installed mISDNuser.i386 1.1.7.2-2_centos5 installed asterisk-chan_misdn.i386 1.4.22-3 trixbox kmod-mISDN-PAE.i686 1.1.7.2-3_centos5.2.6.18_164.11.1.el5 mISDN.i686 1.1.7-27 trixbox mISDN-debuginfo.i686 1.1.7-24 trixboxaddons mISDN-devel.i686 1.1.7-27 trixbox mISDN-devel.i386 1.1.7.2-4_centos5 trixbox28 mISDN-kmod-base.i686 1.1.7.2-1_centos5.2.6.18_128.1.10.el5 mISDN-modules.i686 1.1.7-27.2.6.18_92.1.18.el5 trixbox mISDNuser-debuginfo.i386 1.1.7-15 trixboxaddons mISDNuser-devel.i386 1.1.7.2-2_centos5 trixbox28 All ancient, with many (security) bugs and no longer supported. Asterisk 1.6.0.26-FONCORE-r78, Copyright (C) 1999 - 2010 Digium, Inc. and others. 4, What do you mean with the OS-es were clones ...? Did you create an image of the old Trixbox machine and installed that on the new machine? It means that they are Xen virtual machines, exact bit by bit vm clones so they should have all the same configuration files, run the exact same Xen kernels. What complicates things a bit, and probably the cause of my errors is Xen's PCI passthrough. The only reason why I use something so obsolete like Xen is just this feature otherwise I would be using kvm, vmware, virtualbox or whatever virt technologies but for those you must have vt(d) hardware support and the machine I dealing with here doesn't have this, neither the old one. Right. 5, Lsdadhi (this is on the first, working machine) ### Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS 1 BRIClear (In use) (SWEC: MG2) 2 BRIClear (In use) (SWEC: MG2) 3 BRIHardware-assisted HDLC (In use) ### Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS 4 BRIClear (In use) (SWEC: MG2) 5 BRIClear (In use) (SWEC: MG2) 6 BRIHardware-assisted HDLC (In use) ### Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS RED 7 BRIClear (In use) (SWEC: MG2) RED 8 BRIClear (In use) (SWEC: MG2) RED 9 BRIHardware-assisted HDLC (In use) RED ### Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS 10 BRIClear (In use) (SWEC: MG2) 11 BRIClear (In use) (SWEC: MG2) 12 BRIHardware-assisted HDLC (In use) Ok. 6, Asterisk logs (new machine when it failed) full.4:[Aug 7 12:39:58] WARNING[1654] chan_dahdi.c: Unable to specify channel 1: No such device or address full.4:[Aug 7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to open channel 1:
Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. We are successfully using kamailio / rtpengine with websockets and asterisk 1.8. First question is why are you duplicating registrations within asterisk? Secondly, why are you using websockets in asterisk? Without knowing more about your use case, I'll tell you how we did it. Like I said, kamailio is responsible for our SIP/ws subscribers and registrations. Once within kamailio we simply dispatch traffic to asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng) and that is basically it. No special configuration is needed for asterisk (in fact 1.8 has no support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a kamailio peer and away you go. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Asterisk 13.0.0-beta1 Now Available!
Hello Asterisk, Monday, August 11, 2014, 2:45:00 PM, you wrote: The Asterisk Development Team is pleased to announce the first beta release of Asterisk 13.0.0. This release is available for immediate download at In my living dangerously mode, V13 beta 1 is up and running on my home PBX replacing 12.4 or whatever the most current 12 was. So far it seems exactly the same. -- Ira-- _ -- 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
