[asterisk-users] Record() Cmd and My SQL
HI , Is there Any way is there so that I can store my recordings directly to a database rather storing the same to a file . Thanks in advance . Regards Mahesh -- _ -- 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] Calls stuck in the queue even when ext's are available
Hi.. We are facing a problem that is making the channel to be stuck. we are using asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues and one has 2 agents and the other 5 agents, from last week the second queue's channel is getting stuck, it happened 3 times till now and the problem is calls come into the queue and just the calls will be in the queue and will not ring any agents (static) even when are available..so when i went to the CLI and saw few channels were stuck: SIP/5060-b65171708...@ext-queues:11 Up Queue(8002|t||) SIP/5060-b65110708...@ext-queues:11 Up Queue(8002|t||) SIP/5060-b6515be08...@ext-queues:11 Up Queue(8002|t||) SIP/5060-0854ba808...@ext-queues:11 Up Queue(8002|t||) SIP/5060-08584ad08...@ext-queues:11 Up Queue(8002|t||) Even when i did "soft hangup" it did not hangupso i had to kill the asterisk process and had to restart it..i was researching and found that there is "autofill=yes" option that i am going to try it.please share if you have any thoughts in regards to the queue problem... Thank you very much Regards Sandesh -- _ -- 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] Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever
Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5441. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5442. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 5444. Usually these error stay on the /var/log/messages for ever. I mean they repeat. Is there a problem with these? Thanks -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Calls are not going outside of the network. I had to setup up the subnet of the other side (openvpn client) as the localnet of the Asterisk server for Asterisk to not handle it with NAT or hand shake it with external IP. Thanks, -Bruce On Wed, Sep 22, 2010 at 1:58 PM, Paul Belanger wrote: > On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce wrote: > > Thanks, but Carlos Chavez was right on point. This fixed the problem: > > externip=123.123.123.123 > > localnet=192.168.100.0/255.255.255.0 > > nat=no in each extension. > > > So now I am confused, If you have a VPN setup between sites, why are > calls going outside the VPN? Or do you have remote agents that are > not using a VPN? > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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] func SHARED, how to use?
22.09.2010 16:08, Philipp von Klitzing пишет: > Hi Dmitry! > > > Hello! > And the third hit in my google result is this: > > http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html > > Since I mentioned in my previous message that you will find the answer in > the archive of this list you could have found that even without google. > gmane.org for example has a nice web UI for reading this list. > > I'm sorry, but this is absolutely the same thing I see on voip-info.org. And, I'm too stupid to understand how to use it in dial plan, especially for RTCP statistics. :-( May be this is very-very simple, so nobody understand what I want, if it is absolutely clear... But,could someone provide me example of how to use SHARED with RTCP? Thank you! -- _ -- 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] Recording maximum time and stop on silence
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, September 22, 2010 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recording maximum time and stop on silence All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 AFAIK, #1 is limited only by available disk space, #2 is yes, but you may have to tweak some settings to "get it right" -- _ -- 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] Recording maximum time and stop on silence
All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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- speech to text(Voicemail to text message)
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of amit salunkhe Sent: Wednesday, September 22, 2010 3:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk- speech to text(Voicemail to text message) Dear All Can you let me know is this possible to if we are using Asterisk version 1.4 or 1.6 for incoming voicemail we can send as email in text formta. Means voice mesage converted into text message & send it to resp. email ids. is this possible. If yes. we can do the same with help of Asterisk or we require expertnal application need to isntall/integrate to work for speech to test. Please help me with data to configure. Thanks Amit- FWIW, the current state of Speech-to-text will let you do a 70-95% accurate translation of incoming voicemails depending on clarity/dialect/training. Also depends on language of "native" speakers. For 100% reliability, this still requires Human intervention. -- _ -- 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- speech to text(Voicemail to text message)
Dear All Can you let me know is this possible to if we are using Asterisk version 1.4 or 1.6 for incoming voicemail we can send as email in text formta. Means voice mesage converted into text message & send it to resp. email ids. is this possible. If yes. we can do the same with help of Asterisk or we require expertnal application need to isntall/integrate to work for speech to test. Please help me with data to configure. Thanks Amit-- -- _ -- 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 as a distributed paging system
Hi! > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. > I'd need also to page a subset of all the speakers. Most of the major phone vendors (that are employed by the users of this list) have support for multi-cast of some sort. In recent firmware release notes I have read that SNOM has now also added a feature to feed multicast directly from a phone (and not just play multicast audio on the speaker as long as the phone is not in use). > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. app_page has been around for quite some in Asterisk, and the new Asterisk 1.8 now also adds the channel driver "MulticastRTP". > Is there a way asterisk could be of use, or would I need to bend it > too much? Look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom I have made good experience with MAST for multicasting SNOM phones: http://www.aelius.com/njh/mast/ Philipp -- _ -- 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 as a distributed paging system
With a proper setup and asynchronous dialing, this can be done in a relatively seamless (although not as simple as this indicates) fashion. -- _ -- 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 as a distributed paging system
On Wed, 22 Sep 2010, Matteo Fortini wrote: > I'm building a paging system composed of roughly 10 switches in daisy > chain, with an embedded box with a speaker and a microphone for each > switch. The embedded box runs my software. > > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. I'd > need also to page a subset of all the speakers. > > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. > > I don't own the switches so each site will be assigned an address by > DHCP, that's why I'm using multicast. > > I heard of asterisk and SIP as a possible alternative to my software, > and I'd rather use tested and widely adopted software. > > Is there a way asterisk could be of use, or would I need to bend it too > much? It does this as standard. However, to make it work you need to write some Asterisk dialplan code, and have SIP devices (phones) that can auto-answer and go into speakerphone mode. It also doesn't use multicast, so the number of 'speakers' (phones) you have might be a limitation. I've tested it with 20 without any issues, however as it uses the internal conferernce facilities (meetme), I know that there are people out there using asterisk to host some very large conferernces, so I suspect for your implementation it won't be an issue... Your requirement of any to all reachable, even if the network is broken is tricky - you might end up having an asterisk box at each switch (which I assume is an Ethernet switch) although then you'll have problems with the SIP device registrations. Gordon -- _ -- 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] Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6
Hello I recently heard this should be possible. Has anyone experience with this? Thanks! -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce wrote: > Thanks, but Carlos Chavez was right on point. This fixed the problem: > externip=123.123.123.123 > localnet=192.168.100.0/255.255.255.0 > nat=no in each extension. > So now I am confused, If you have a VPN setup between sites, why are calls going outside the VPN? Or do you have remote agents that are not using a VPN? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks, but Carlos Chavez was right on point. This fixed the problem: externip=123.123.123.123 localnet=192.168.100.0/255.255.255.0 nat=no in each extension. Maybe combination of both or only the localnet just fixed it. Thanks, Bruce On Wed, Sep 22, 2010 at 1:35 PM, Steve Edwards wrote: > Un-top-posting... > > > On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce > wrote: > > > Any feed back is appreciated. > > > On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger < > paul.belan...@polybeacon.com> wrote: > > > Then configure you endpoints to use the 192.168.100.0/24 network. This > > is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending > > the INVITE message. > > On Wed, 22 Sep 2010, bruce bruce wrote: > > > I don't think it's an endpoint issue. I think the SIP packet headers get > > over-written by the tunnel (openvpn) protocol. > > Would wireshark shed some light? > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Un-top-posting... > On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce > wrote: > > Any feed back is appreciated. > On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger > wrote: > Then configure you endpoints to use the 192.168.100.0/24 network. This > is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending > the INVITE message. On Wed, 22 Sep 2010, bruce bruce wrote: > I don't think it's an endpoint issue. I think the SIP packet headers get > over-written by the tunnel (openvpn) protocol. Would wireshark shed some light? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] http://www.asterisk.org/downloads naming schema
On Wed, Sep 22, 2010 at 09:50:00AM -0700, Steve Edwards wrote:
>
> Still, for scripting and portability, I'd recommend specifying the
> "decompressor" and using the long option form:
>
> tar\
> --list\
> --[un]gzip\
> --file\
> asterisk-1.4-current.tar.gz
Those are GNU tar options, which {Free,Net,Open}BSD won't like.
--
Barry
--
_
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks for that Carlos. I am playing with that right now. What do you suggest localnet should say? Server A = OpenVPN Server: localnet=127.0.01 localnet=192.168.100.0/255.255.255.0 Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client) Server A doesn't have any localnet other than the loop back and then a Vnet to internet (public ip address). Thanks, Bruce On Wed, Sep 22, 2010 at 11:36 AM, Carlos Chavez wrote: > Do you have a localnet statement in your sip.conf? That and using > nat=no will make sure Asterisk does not replace the IP address in the > Invite. > > On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote: > > Hi Everyone, > > > > > > I have setup an OpenVPN tunnel between Server A (running Asterisk) and > > Server B suppling it's SIP Phones with DHCP pool of IPs. > > > > > > So, the tunnel is established nicely and everyone can ping others. > > "sip show peers" shows the local subnet of the SIP Phones registered > > (192.168.100.0/24). > > > > > > But there is the old bad one-way audio. Calls also drop after few > > seconds. In the SIP debug I can see that asterisk uses it's external > > public IP address to communicate to endpoints that are known to it as > > the 192.168.100.0/24 endpoints and the endpoints identify themselves > > with the OpenVPN tunnel IP address scheme in one part of the sip > > handshake. How can this be fixed? After all, with the OpenVPN this > > should all look like an internal network to Asterisk. > > > > > > I have added my comments followed by # to lines below that are > > problematic. > > > > > > <--- SIP read from UDP:192.168.100.5:5060 --->#This line is good > > as it uses the local DHCP supplied network address scheme > > INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we > > inviting Ext. 203 with it's OpenVPN IP while it's on the same network > > of 192.168.50.0/24 as 202? > > Via: SIP/2.0/UDP > > 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 > Max-Forwards: 70 > > From: "SIP Phone - Ext. 202" ;tag=6d6f8c4226 > >#BAD line again. Should be > > SIP:2...@192.168.100.6 > > To: "203" #Bad again > > Call-ID: 43af67a634e06e75 > > CSeq: 32058 INVITE > > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, > > PRACK, SUBSCRIBE, INFO > > Allow-Events: talk, hold, conference, LocalModeStatus > > Contact: "SIP Phone - Ext. 202" > > ; > > +sip.instance="" > > Supported: gruu, path, timer, 100rel, replaces > > User-Agent: Aastra 55i/2.5.2.1500 > > Content-Type: application/sdp > > Content-Length: 594 > > > > > > Basically the phones should only send with FROM their local > > 192.168.100.0/24 address and Asterisk should only send ANSWER and ACK > > back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 > > (which is the openvpn client ip). > > > > > > Once above is fixed, I think all the audio and call cut will go away. > > I hate to use a sip proxy in this situation since I already have an > > openvpn connection. > > > > > > Any feed back is appreciated. > > > > > > Thanks, > > -- > > _ > > -- 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 > -- > Carlos Chavez > Director de Tecnología > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > -- > _ > -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
I don't think it's an endpoint issue. I think the SIP packet headers get over-written by the tunnel (openvpn) protocol. Thanks, Bruce On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger wrote: > On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote: > > Any feed back is appreciated. > > > Then configure you endpoints to use the 192.168.100.0/24 network. > This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is > sending the INVITE message. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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] http://www.asterisk.org/downloads naming schema
On 10-09-22 11:45 AM, Klaus Darilion wrote: > Hi! > > Since some time the download of the newest Asterisk does not contains > the version number anymore, but is just called "asterisk-1.4-current.tar.gz" > > This gives me a tarball where I do not know the version without looking > into the tarball. > > Thus, IMO it would be very useful to switch back to old schema war the > download contained the version number. I don't understand really. The downloads.asterisk.org site contains the current version in the pub/telephony/asterisk/ directory, and there is a symlink to the current version which is named asterisk-1.4-current. On the Downloads page on asterisk.org we have the link setup to asterisk-1.4-current (and 1.6.2-current, etc.) but that again is just the symlink to the currently available version. The Downloads page is also updated with text in the table with the currently available version, such as 1.4.36 or 1.6.2.13, etc, so I'm not sure what you're asking to be changed. We've been doing it like this for quite some time (in the timeframe of years, to my knowledge). Thanks! Leif. -- _ -- 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] http://www.asterisk.org/downloads naming schema
>> On Wed, 22 Sep 2010, Jose P. Espinal wrote: >> >>> If you are using a script you could get the version with something like: >>> >>> tar -tf asterisk-1.4-current.tar.gz | head -n1 > On 09/22/2010 11:20 AM, Steve Edwards wrote: >> You need a '-z' in there. On Wed, 22 Sep 2010, Kevin P. Fleming wrote: > Modern versions of 'tar' auto-detect gzip and bzip compression :-) Gee. Who knew :) CentOS 4.8 (tar 1.14) doesn't. CentOS 5.5 (tar 1.15.1) does. Still, for scripting and portability, I'd recommend specifying the "decompressor" and using the long option form: tar\ --list\ --[un]gzip\ --file\ asterisk-1.4-current.tar.gz -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 as a distributed paging system
I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm currently using some software I wrote which sends voice over multicast RTP and coordinates all the sites with multicast messages. I don't own the switches so each site will be assigned an address by DHCP, that's why I'm using multicast. I heard of asterisk and SIP as a possible alternative to my software, and I'd rather use tested and widely adopted software. Is there a way asterisk could be of use, or would I need to bend it too much? Thank you in advance, Matteo -- _ -- 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] http://www.asterisk.org/downloads naming schema
Oh, my bad. It my box there might be some defaults predefined, as it did not yield any errors. Steve Edwards wrote: > On Wed, 22 Sep 2010, Jose P. Espinal wrote: > >> If you are using a script you could get the version with something like: >> >> tar -tf asterisk-1.4-current.tar.gz | head -n1 > > You need a '-z' in there. > -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] http://www.asterisk.org/downloads naming schema
On 09/22/2010 11:20 AM, Steve Edwards wrote: > On Wed, 22 Sep 2010, Jose P. Espinal wrote: > >> If you are using a script you could get the version with something like: >> >> tar -tf asterisk-1.4-current.tar.gz | head -n1 > > You need a '-z' in there. Modern versions of 'tar' auto-detect gzip and bzip compression :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] http://www.asterisk.org/downloads naming schema
On Wed, 22 Sep 2010, Jose P. Espinal wrote: > If you are using a script you could get the version with something like: > > tar -tf asterisk-1.4-current.tar.gz | head -n1 You need a '-z' in there. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Solving the CDR mess of attended transfers
Is there a documentation about the CEL format? l. 2010/9/22 Steve Murphy > > CEL was my answer, built on the channel event goodness that Russell. It's > now in 1.8; but it > lacks a converter to CDRs. You *could* just use the string of events coming > out of CEL, but... > I'd love to see your SQL statements to pull things together! > > I had begun writing a CEL->CDR converter, but got laid off before I could > finish it. > It makes a good start toward a finished package. Long ago (what, almost 2 > years now?) > I proposed two methods of generating CDR's. One was 'simple', the other > 'Complex", or "Leg Based". > > Since then, I refined the doc to just 'Simple', and outlined with some > examples how it would/should work. > The doc still needs to be cleaned up, but you may make sense of it. > > The trouble with CDRs is that no two shops can agree on a CDR standard that > involves transfers, parks, etc. > Beyond the "start", "answer", and "end" times, and some fundamental data > about the call (source, dest, > responsible party, etc.) There isn't much unity about what timepoints need > to be represented, etc. And I'd seen > so few implementations, I couldn't judge a good way to generalize the CDR > converter. > > So, I challenge everyone to look at my simple CDR definition, and see it > would possible for you to adapt it > (perhaps via a mapping from it to your desired conflagration/configuration. > > To look at the doc, do "svn co > http://svn.digium.com/svn/team/murf/asterisk-RFCs and look at the > document in there (I have a few different formats, the .docx is the > source). > > It's been in flux. Just the first few examples are accurate. Let me know > what you think. > > murf > > > > -- > Steve Murphy > ParseTree Corp > > > -- > _ > -- 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 > -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] http://www.asterisk.org/downloads naming schema
Hi Klaus, If you are using a script you could get the version with something like: tar -tf asterisk-1.4-current.tar.gz | head -n1 Regards, Klaus Darilion wrote: > Hi! > > Since some time the download of the newest Asterisk does not contains > the version number anymore, but is just called "asterisk-1.4-current.tar.gz" > > This gives me a tarball where I do not know the version without looking > into the tarball. > > Thus, IMO it would be very useful to switch back to old schema war the > download contained the version number. > > Thanks > Klaus > -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] http://www.asterisk.org/downloads naming schema
On 09/22/2010 10:55 AM, Steve Howes wrote: > On 22 Sep 2010, at 16:45, Klaus Darilion wrote: >> Since some time the download of the newest Asterisk does not contains >> the version number anymore, but is just called "asterisk-1.4-current.tar.gz" >> >> This gives me a tarball where I do not know the version without looking >> into the tarball. >> >> Thus, IMO it would be very useful to switch back to old schema war the >> download contained the version number. > > http://downloads.asterisk.org/pub/telephony/asterisk/ You can either download the 'newest version', or you can download a specific version. If your tell your browser to download asterisk-1-4.current.tar.gz, the server can't tell your browser to actually give that file a different name after downloading it... although it's possible we could come up with some creative HTTP redirect mechanism that redirected your browser to the version-numbered filename, instead of using a symbolic link on the filesystem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] http://www.asterisk.org/downloads naming schema
On Wed, Sep 22, 2010 at 11:45 AM, Klaus Darilion wrote: > This gives me a tarball where I do not know the version without looking > into the tarball. > Should be simple to do, since http://www.asterisk.org/downloads/asterisk/releases/asterisk-1.8.0-betaX.tar.gz currently redirects to the proper download. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] http://www.asterisk.org/downloads naming schema
Klaus Darilion wrote: > Hi! > > Since some time the download of the newest Asterisk does not contains > the version number anymore, but is just called "asterisk-1.4-current.tar.gz" > > This gives me a tarball where I do not know the version without looking > into the tarball. > > Thus, IMO it would be very useful to switch back to old schema war the > download contained the version number. > > Thanks > Klaus > Its normally just a symbolic link to the current version. If you untar the archive the directory name will represent the actual software version. If you want to switch back to an old version then you are best off keeping the uncompresssed and already compiled version anyway as you know for sure you are reinstalling the older version exactly as it was before. -- _ -- 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] http://www.asterisk.org/downloads naming schema
On 22 Sep 2010, at 16:45, Klaus Darilion wrote: > Since some time the download of the newest Asterisk does not contains > the version number anymore, but is just called "asterisk-1.4-current.tar.gz" > > This gives me a tarball where I do not know the version without looking > into the tarball. > > Thus, IMO it would be very useful to switch back to old schema war the > download contained the version number. http://downloads.asterisk.org/pub/telephony/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
[asterisk-users] http://www.asterisk.org/downloads naming schema
Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Do you have a localnet statement in your sip.conf? That and using nat=no will make sure Asterisk does not replace the IP address in the Invite. On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote: > Hi Everyone, > > > I have setup an OpenVPN tunnel between Server A (running Asterisk) and > Server B suppling it's SIP Phones with DHCP pool of IPs. > > > So, the tunnel is established nicely and everyone can ping others. > "sip show peers" shows the local subnet of the SIP Phones registered > (192.168.100.0/24). > > > But there is the old bad one-way audio. Calls also drop after few > seconds. In the SIP debug I can see that asterisk uses it's external > public IP address to communicate to endpoints that are known to it as > the 192.168.100.0/24 endpoints and the endpoints identify themselves > with the OpenVPN tunnel IP address scheme in one part of the sip > handshake. How can this be fixed? After all, with the OpenVPN this > should all look like an internal network to Asterisk. > > > I have added my comments followed by # to lines below that are > problematic. > > > <--- SIP read from UDP:192.168.100.5:5060 --->#This line is good > as it uses the local DHCP supplied network address scheme > INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we > inviting Ext. 203 with it's OpenVPN IP while it's on the same network > of 192.168.50.0/24 as 202? > Via: SIP/2.0/UDP > 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 > Max-Forwards: 70 > From: "SIP Phone - Ext. 202" ;tag=6d6f8c4226 >#BAD line again. Should be SIP:2...@192.168.100.6 > To: "203" #Bad again > Call-ID: 43af67a634e06e75 > CSeq: 32058 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, > PRACK, SUBSCRIBE, INFO > Allow-Events: talk, hold, conference, LocalModeStatus > Contact: "SIP Phone - Ext. 202" > ; > +sip.instance="" > Supported: gruu, path, timer, 100rel, replaces > User-Agent: Aastra 55i/2.5.2.1500 > Content-Type: application/sdp > Content-Length: 594 > > > Basically the phones should only send with FROM their local > 192.168.100.0/24 address and Asterisk should only send ANSWER and ACK > back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 > (which is the openvpn client ip). > > > Once above is fixed, I think all the audio and call cut will go away. > I hate to use a sip proxy in this situation since I already have an > openvpn connection. > > > Any feed back is appreciated. > > > Thanks, > -- > _ > -- 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 -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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 T38
That's probably what I'm going to have to do. Thanks. > I suppose that merely removing ATA and asterisk from the middle, and > plugging a pots line into a fax machine is out of the question. > > -- _ -- 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] TLS re-negotiation attack on SIP/TLS of Asterisk?
Hi all, i read about the TLS-RENEGOTIATION vulnerability: http://www.educatedguesswork.org/2009/11/understanding_the_tls_renegoti.html http://www.sslshopper.com/article-ssl-and-tls-renegotiation-vulnerability-discovered.html www.phonefactor.com/sslgapdocs/Renegotiating_TLS.pdf Does the Asterisk 1.6/1.8 SIP/TLS implementation suffer from the TLS Renegotiation vulnerability or the TLS-renegotiation it's disabled by default, in how OpenSSL is used? Fabio Pietrosanti -- _ -- 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 T38
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett wrote: > In the simplest terms I can think of, I'm going to describe what I want to > do and I want to know if it's possible in the current version of asterisk. > > Can I take a T38 call from an ATA, convert that back to analog and have > asterisk screech that out on a POTS line to a remote fax machine. Would it > work? > > And could I receive an incoming fax the same way? I suppose that merely removing ATA and asterisk from the middle, and plugging a pots line into a fax machine is out of the question. Sounds like you want a T.38 gateway. Not built into asterisk, but some people have tried patching. Search the archives for T.38 gateway. -- _ -- 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 T38
On 09/22/2010 09:00 AM, Adam Moffett wrote: > In the simplest terms I can think of, I'm going to describe what I want > to do and I want to know if it's possible in the current version of > asterisk. > > Can I take a T38 call from an ATA, convert that back to analog and have > asterisk screech that out on a POTS line to a remote fax machine. Would > it work? > > And could I receive an incoming fax the same way? This is called T.38 gateway mode, and it's not available in any Asterisk releases yet. This has been discussed quite often on this mailing list, though, so a Google search of the list archives would give you pointers to the methods you can use today to achieve this. Asterisk 1.8 was just enhanced to provide some new APIs that will be necessary for seamless implementation of T.38 gateway mode, and we expect that work on that will occur in the very near future. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Costa Rica Hangup Detection
On Wed, Sep 22, 2010 at 10:05 AM, Gustavo A. Gonzalez wrote: > Hi all! I'm configuring a digium tdm card in Costa Rica, every things > works well, but calls don't hangup. I've tested setting up progzone=br > but dont work. Thanks for any help. > Does you telco provide a disconnect tone? Most don't. Your best to record the call, and analyst the tone, you can then update indications.conf. The next best thing is to implement timeouts within your dialplans. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Costa Rica Hangup Detection
Hi all! I'm configuring a digium tdm card in Costa Rica, every things works well, but calls don't hangup. I've tested setting up progzone=br but dont work. Thanks for any help. Cheers! -- Gustavo A. González Dto. Telefonía VoIP Despegar.com 54 (11) 5032-3500 ext. 3512 -- _ -- 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 T38
In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me about alternatives to faxing. I can't take the fax machine away from the end user, they don't want to hear about it. I either need to make it work or tell them to get a POTS line. <>-- _ -- 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] T38 and codecs negotiation
On Wed, Sep 22, 2010 at 7:58 AM, federico cabiddu wrote: > This did the trick for me but I don't know the implications of such change > and if it is correct to manage it this way. > It might we worth following up with a developer on #asterisk-dev, then submitting your patch to https://issues.asterisk.org -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Can't cross compile asterisk 1.6.2.13 on arm using ltib
On Wed, Sep 22, 2010 at 9:21 AM, IMS wrote: > Do you have any ideas of the problem ? config.log don't give me more > explanations. > Attach your config.log so we can see what is going on. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Cross compile Asterisk for mipsel-linux
On Wed, Sep 22, 2010 at 5:42 AM, Nikhil wrote: > Anyone knows how to do cross compile asterisk 1.6.2.13 using > mipsel linux.? > $ ./configure --help Will output the flags you need to set. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote: > Any feed back is appreciated. > Then configure you endpoints to use the 192.168.100.0/24 network. This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending the INVITE message. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks for the feedback. I thought about that but it's not an option for me right now. Any other ways folks? Thanks On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West wrote: > On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote: > >I have setup an OpenVPN tunnel between Server A (running Asterisk) and > >Server B suppling it's SIP Phones with DHCP pool of IPs. > > Have you considered running Asterisk on Server B as well, and using IAX > to trunk between them? This is working well for me. > > Roger > > -- > _ > -- 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] Solving the CDR mess of attended transfers
A few corrections! On Tue, Sep 21, 2010 at 6:32 PM, Steve Murphy wrote: > > > On Tue, Sep 7, 2010 at 5:36 PM, Fabiano Carlos Heringer < > b...@grupoheringer.com.br> wrote: > >> Em 07/09/2010 17:15, Miguel Molina escreveu: >> >> El 07/09/10 14:49, Fabiano Carlos Heringer escribió: >> >> Is there a way to solve the mess on CDR caused by CDR Transfer? anyway, by >> paid support, no paid, or another way... Im going crazy about this. My boss >> is really furious because he don´t understand nothing on the CDR. >> >> I tried the 1.6.2.11, Asterisk 1.8 beta, and everything still the same. >> >> Any solution? >> >> Thanks! >> >> Hi >> >> Some quick measures: >> >> 1. Enable unanswered=yes on cdr.conf and try to see if it helps you with >> the CDR. >> 2. Try using CEL (Channel Event Logging) in 1.8-beta and try to see if >> that helps in a definite way. >> >> Cheers, >> >> -- >> Ing. Miguel Molina >> Grupo de Tecnología >> Millenium Phone Center >> >> Hi, will make this change on my cdr.conf >> >> About CEL on asterisk 1.8 i tried some test on my test server, he really >> logs each event on log, but i did not understood how he will work on a user >> view (most simple). It´s possible to log this events on a database such >> mysql? >> >> Thanks! >> > > Sorry for the delay, things have been busy here. > > Yes, there are problems with the existing CDR interface, mostly historical, > because as Asterisk > grew, the CDR system became obsolete. There were attempts to make it work, > but structurally > and architecturally, it was just not going to work. > > CEL was my answer, built on the channel event goodness that Russell. It's > now in 1.8; but it > Uh, that Russell *wrote*. > lacks a converter to CDRs. You *could* just use the string of events coming > out of CEL, but... > I'd love to see your SQL statements to pull things together! > > I had begun writing a CEL->CDR converter, but got laid off before I could > finish it. > It makes a good start toward a finished package. Long ago (what, almost 2 > years now?) > I proposed two methods of generating CDR's. One was 'simple', the other > 'Complex", or "Leg Based". > > Since then, I refined the doc to just 'Simple', and outlined with some > examples how it would/should work. > The doc still needs to be cleaned up, but you may make sense of it. > > The trouble with CDRs is that no two shops can agree on a CDR standard that > involves transfers, parks, etc. > Beyond the "start", "answer", and "end" times, and some fundamental data > about the call (source, dest, > responsible party, etc.) There isn't much unity about what timepoints need > to be represented, etc. And I'd seen > so few implementations, I couldn't judge a good way to generalize the CDR > converter. > > So, I challenge everyone to look at my simple CDR definition, and see it > would possible for you to adapt it > (perhaps via a mapping from it to your desired conflagration/configuration. > > To look at the doc, do "svn co > http://svn.digium.com/svn/team/murf/asterisk-RFCs and look at the > document in there (I have a few different formats, the .docx is the > source). > Sorry, the URL is http://svn.digium.com/svn/asterisk/team/murf/RFCs > > It's been in flux. Just the first few examples are accurate. Let me know > what you think. > > murf > > > > -- > Steve Murphy > ParseTree Corp > > -- Steve Murphy ParseTree Corp -- _ -- 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] Can't cross compile asterisk 1.6.2.13 on arm using ltib
Hi, I can cross compile asterisk 1.4.21 on arm (imx27) using ltib I want to cross compile the new version 1.6.2.13 but there is an error when I execute the commands : ./configure --build=i686-pc-linux-gnu --host=arm make menuselect The configure seems ok, I have the result info : *configure: Package configured for: configure: OS type : none configure: Host CPU : arm configure: build-cpu:vendor:os: i686 : pc : linux-gnu : configure: host-cpu:vendor:os: arm : unknown : none : configure: Cross Compilation = YES * But when I try to execute make menuselect I have the message : *CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts make[1]: Entering directory `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect' configure: error: in `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect': configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. make[1]: *** [makeopts] Error 1 make[1]: Leaving directory `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect' make: *** [menuselect/makeopts] Error 2* Do you have any ideas of the problem ? config.log don't give me more explanations. With google i found the problem should be corrected from the revision 268052 (Build menuselect with the build environment's compiler, not the host (target)'s compiler) here : http://svnview.digium.com/svn/asterisk/branches/1.6.2?view=revision&revision=268052 Thanks for your help. Sebastien -- _ -- 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] Unable to open vm-INBOXs
>-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >Jonas Kellens >Sent: Wednesday, September 22, 2010 9:04 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] Unable to open vm-INBOXs > >On 09/22/2010 02:45 PM, Philipp von Klitzing wrote: >> .slin is not .wav >> > >Other files that are also in wav format play without any problem : > >[Sep 22 15:02:35] -- Playing >'vm-youhave.slin' (language 'nl') > >[r...@asterisk16 asterisk-1.6.2.10]# ls -l >/var/lib/asterisk/sounds/nl/ total 388 drwxr-xr-x 2 root root >4096 Sep 22 11:25 digits >-rw-r--r-- 1 root root 66124 Sep 22 11:10 vm-helpexit.wav >-rw-r--r-- 1 root root44 Sep 22 14:19 vm-INBOXs.wav >-rw-r--r-- 1 root root 16844 Sep 22 12:47 vm-INBOX.wav >-rw-r--r-- 1 root root 37004 Sep 22 10:58 vm-incorrect.wav >-rw-r--r-- 1 root root 26764 Sep 22 12:47 vm-messages.wav >-rw-r--r-- 1 root root 23564 Sep 22 12:54 vm-message.wav >-rw-r--r-- 1 root root 12364 Sep 22 11:06 vm-no.wav >-rw-r--r-- 1 root root 19404 Sep 22 14:19 vm-Olds.wav >-rw-r--r-- 1 root root 17164 Sep 22 14:20 vm-Old.wav >-rw-r--r-- 1 root root 27404 Sep 22 12:49 vm-onefor.wav >-rw-r--r-- 1 root root 31884 Sep 22 10:57 vm-password.wav >-rw-r--r-- 1 root root 25164 Sep 22 11:04 vm-youhave.wav > > > >Jonas. > >-- Well, I think I see the problem now that you've shown a directory listing. The file in question is a mere 44 bytes. That is almost certainly not right. - Brad -- _ -- 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] Unable to open vm-INBOXs
On 09/22/2010 02:45 PM, Philipp von Klitzing wrote: > .slin is not .wav > Other files that are also in wav format play without any problem : [Sep 22 15:02:35] -- Playing 'vm-youhave.slin' (language 'nl') [r...@asterisk16 asterisk-1.6.2.10]# ls -l /var/lib/asterisk/sounds/nl/ total 388 drwxr-xr-x 2 root root 4096 Sep 22 11:25 digits -rw-r--r-- 1 root root 66124 Sep 22 11:10 vm-helpexit.wav -rw-r--r-- 1 root root44 Sep 22 14:19 vm-INBOXs.wav -rw-r--r-- 1 root root 16844 Sep 22 12:47 vm-INBOX.wav -rw-r--r-- 1 root root 37004 Sep 22 10:58 vm-incorrect.wav -rw-r--r-- 1 root root 26764 Sep 22 12:47 vm-messages.wav -rw-r--r-- 1 root root 23564 Sep 22 12:54 vm-message.wav -rw-r--r-- 1 root root 12364 Sep 22 11:06 vm-no.wav -rw-r--r-- 1 root root 19404 Sep 22 14:19 vm-Olds.wav -rw-r--r-- 1 root root 17164 Sep 22 14:20 vm-Old.wav -rw-r--r-- 1 root root 27404 Sep 22 12:49 vm-onefor.wav -rw-r--r-- 1 root root 31884 Sep 22 10:57 vm-password.wav -rw-r--r-- 1 root root 25164 Sep 22 11:04 vm-youhave.wav 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] Unable to open vm-INBOXs
>-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >Jonas Kellens >Sent: Wednesday, September 22, 2010 8:26 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] Unable to open vm-INBOXs > >This is what happens : > >[Sep 22 14:22:42] -- Playing 'vm-INBOXs.slin' >(language 'nl') >[Sep 22 14:22:42] == Spawn extension (from-TEST, 1001, 5) exited >non-zero on 'SIP/test6-0008' > > >Asterisk ends the conversation because the file 'vm-INBOXs' >does not exist. > >But the file is present : > >[r...@asterisk16 asterisk-1.6.2.10]# locate vm-INBOXs >/var/lib/asterisk/sounds/nl/vm-INBOXs.wav > Well this is completely different from what you originally posted... Anyway, what is the output of 'core show file formats'? It sounds like you're missing a format_XXX.so (perhaps unselected in menuselect?) and so the channel is falling back to trying to find a signed linear file (which, at least in name, you don't have). - Brad -- _ -- 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] Unable to open vm-INBOXs
.slin is not .wav -- _ -- 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] Unable to open vm-INBOXs
On 09/22/2010 01:38 PM, Watkins, Bradley wrote: > This is indicative that you have set the channel's language to something > that expects there to be a singular and plural version of the 'new' (as > in 'one new message' versus 'five new messages') sound. > > According to the code, that includes Dutch, Spanish, Portuguese and > Greek. > > If you have one of these set as your language (I'm guessing Dutch), then > the sound file set you have is incomplete. > > Regards, > - Brad > This is what happens : [Sep 22 14:22:42] -- Playing 'vm-INBOXs.slin' (language 'nl') [Sep 22 14:22:42] == Spawn extension (from-TEST, 1001, 5) exited non-zero on 'SIP/test6-0008' Asterisk ends the conversation because the file 'vm-INBOXs' does not exist. But the file is present : [r...@asterisk16 asterisk-1.6.2.10]# locate vm-INBOXs /var/lib/asterisk/sounds/nl/vm-INBOXs.wav 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] func SHARED, how to use?
Hi Dmitry! > > Have you considered using Google (or your favourite search engine)? > > Shure, I searched and find nothing. > > The search terms "C" will surely help you, and in > > fact point you to the very archive of this mailing list. Don't know where this quote comes from, but "C" is absolutely not what I wrote. Instead the terms that I mentioned were: asterisk function shared And the third hit in my google result is this: http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html Since I mentioned in my previous message that you will find the answer in the archive of this list you could have found that even without google. gmane.org for example has a nice web UI for reading this list. Philipp -- _ -- 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] T38 and codecs negotiation
Hi,
I'm working with asterisk 1.4.35 and found an issue regarding codecs
negotiation when T38 is enabled (t38pt_udptl=yes).
In particular if the INVITE sdp contains no allowed codec the call is not
rejected with "488 - Not acceptable here" but it goes through and the 200 OK
SDP is as follows:
v=0
o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio x RTP/AVP
a=silenceSupp:off - - - -
a=sendrecv
or
v=0
o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio x RTP/AVP 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=silenceSupp:off - - - -
a=sendrecv
if in the originating INVITE there was the a line for telephone-event
mapping.
Looking chan_sip.c I understood that the problem is related to t38
capabilities and in particular at row 5636:
if (!newjointcapability) {
/* If T.38 was not negotiated either, totally bail out... */
if (!p->t38.jointcapability || !udptlportno) {
ast_log(LOG_NOTICE, "No compatible codecs, not accepting
this offer!\n");
/* Do NOT Change current setting */
return -1;
} else {
if (option_debug > 2)
ast_log(LOG_DEBUG, "Have T.38 but no audio codecs,
accepting offer anyway\n");
}
}
As I understand if t38 is globally enabled p->t38.jointcapability
and udptlportno are always true even so the call is never rejected.
As this behavior caused me some problems with a customer I modified, for the
moment, the line 5638 as follows:
if (!p->t38.jointcapability || !udptlportno || p->t38.state == T38_DISABLED)
cause if I understand well the code if there is no fax request in the INVITE
SDP p->t38.state is set to T38_DISABLED.
This did the trick for me but I don't know the implications of such change
and if it is correct to manage it this way.
Kind regards,
Federico Cabiddu
--
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Re: [asterisk-users] Unable to open vm-INBOXs
This is indicative that you have set the channel's language to something that expects there to be a singular and plural version of the 'new' (as in 'one new message' versus 'five new messages') sound. According to the code, that includes Dutch, Spanish, Portuguese and Greek. If you have one of these set as your language (I'm guessing Dutch), then the sound file set you have is incomplete. Regards, - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, September 22, 2010 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Unable to open vm-INBOXs Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile on my system. Can't imagine this file is in the "extra-sounds" category ?! Al the other voicemail-related sound files are present. Kind regards, 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] func SHARED, how to use?
22.09.2010 15:12, Andrea Cristofanini пишет: >> Could you, please, give me link ? :-) >> > Google is not difficult to use... BTW > http://www.voip-info.org/wiki/view/Asterisk+func+shared > > There is no example here! I already wrote about this... -- _ -- 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] func SHARED, how to use?
> Could you, please, give me link ? :-) Google is not difficult to use... BTW http://www.voip-info.org/wiki/view/Asterisk+func+shared -- --- Andrea Cristofanini Chief Technical Officer ZeroZero39 srlTel: +39 02 61294759 Viale Brianza, 20 Fax: +39 02 87365813 20092 Cinisello Balsamo (MI) Mob:+39 329 1871756 web: www.zerozero39.it PGP Key: http://www.zerozero39.it/cristofanini.asc -- _ -- 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] func SHARED, how to use?
22.09.2010 14:50, Philipp von Klitzing пишет: > Hi! > > >> I see. I want to use SHARED function! >> Do you have example how to >> "to export them to the local call leg/channel "? >> > Have you considered using Google (or your favourite search engine)? > Shure, I searched and find nothing. > The search terms "C" will surely help you, and in > fact point you to the very archive of this mailing list. > > Could you, please, give me link ? :-) -- _ -- 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] func SHARED, how to use?
Hi! > I see. I want to use SHARED function! > Do you have example how to > "to export them to the local call leg/channel "? Have you considered using Google (or your favourite search engine)? The search terms "asterisk function shared" will surely help you, and in fact point you to the very archive of this mailing list. Philipp -- _ -- 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] Unable to open vm-INBOXs
On Wed, Sep 22, 2010 at 12:32:21PM +0200, Jonas Kellens wrote: >[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: >File vm-INBOXs does not exist in any format >[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable >to open vm-INBOXs (format 0x8 (alaw)): No such file or directory >I do not find this particular soundfile on my system. How are you invoking it? That terminal "s" on the filename looks rather unexpected. R -- _ -- 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] Unable to open vm-INBOXs
Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile on my system. Can't imagine this file is in the "extra-sounds" category ?! Al the other voicemail-related sound files are present. Kind regards, 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
[asterisk-users] Cross compile Asterisk for mipsel-linux
Hi Anyone knows how to do cross compile asterisk 1.6.2.13 using mipsel linux.? Thanks -- _ -- 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] func SHARED, how to use?
21.09.2010 18:57, Philipp von Klitzing пишет: > Hi! > > >> Could somebody tell me how to use SHARED function? >> > http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared > > There are no examples there :-( >> I want to get RTCP stats from SIP, but current channel is DAHDI. >> How can I get SIP channel? >> > If you have one DADHI and one SIP channel bridged together, then only for > the SIP channel you will be able to retrieve rtcp data. Depending on > wether that SIP channel is the first (local) or the second (outbound or > remote) call leg you will need to follow the approach described here: > > http://www.voip-info.org/wiki/index.php?page=Asterisk+rtcp > > Quote: > "Use the M option of Dial() if you would like to get the codec > (audionativeformat) of the remote call leg/channel and similar data. > Those are not available anymore during the hangup phase (h extension), > however you can store them directly in the CDR system, or use the SHARED > function to export them to the local call leg/channel." > > I see. I want to use SHARED function! Do you have example how to "to export them to the local call leg/channel "? Thank you! -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote: >I have setup an OpenVPN tunnel between Server A (running Asterisk) and >Server B suppling it's SIP Phones with DHCP pool of IPs. Have you considered running Asterisk on Server B as well, and using IAX to trunk between them? This is working well for me. Roger -- _ -- 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
