Re: [asterisk-users] 1000 analogue lines with asterisk

That would be the expensive route. The inexpensive route would be to buy FXS ethernet gateways, like this: http://www.voipsupply.com/grandstream-gxw4248. You could then get by with a single reasonably sized asterisk box (probably two setup as HA) and no need for expensive cards or complex

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar wrote: > On 2016-02-17 15:32, Richard Mudgett wrote: > >> On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar >> wrote: >> >> Hi everyone. >>> >>> We have an Asterisk server running Debian Squeeze, with

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On 2016-02-17 15:32, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code

Re: [asterisk-users] 1000 analogue lines with asterisk

+1 spending money to get that many fxs ports is going to negate any savings of reusing analog phones instead of buying ip phones 1000 analog ports sounds like hell and if it was me I would be embarrassed to have a setup like that tied to my name if I was a consultant etc. Someone will come in

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: > Hi everyone. > > We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 > (basically, the Debian Stable version for Squeeze, but with some minor > source code changes specific to our site). We're

[asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run

Re: [asterisk-users] res_pjsip trunk between Asterisk servers

On Wed, Feb 17, 2016 at 3:02 PM, Anthony Critelli wrote: > George, thanks so much for the help on this. The wizards did the trick! > > ​Cool! Feedback is always welcomed!​ > Sincerely, > > Anthony Critelli > B.S. Applied Networking and Systems Administration, 2014 >

Re: [asterisk-users] res_pjsip trunk between Asterisk servers

George, thanks so much for the help on this. The wizards did the trick! Sincerely, Anthony Critelli B.S. Applied Networking and Systems Administration, 2014 www.acritelli.com (845) 283-4117 On Mon, Feb 8, 2016 at 10:08 PM, George Joseph wrote: > > > On Mon, Feb 8,

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say

Re: [asterisk-users] 1000 analogue lines with asterisk

Thank you all, I will look at the options and work on them. Regards On Wed, Feb 17, 2016 at 10:22 AM, A J Stiles wrote: > On Wednesday 17 Feb 2016, Goke Aruna wrote: > > Hello all, > > Can someone recommend what hardware to use for a 1000 analogue line > >

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Wow. Incredible. That worked. The backslash is important there; I kept trying with no backslash and followed the instructions in pjsip_wizard.conf.sample (in configs/samples) and it says we have to say transport=tcp ; the only example however talks about ipv4. Is this documented somewhere and I

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am

Re: [asterisk-users] Error making dahdi linux compete 2.11.0

On Mon, Feb 15, 2016 at 05:28:14PM +0200, Tzafrir Cohen wrote: > On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote: > > Getting the some errors making dahdi 2.11.0. > > > > Seems same as listed here > > http://forums.asterisk.org/viewtopic.php?f=1=96455 > > > > In that link they say

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

I made some progress. The first thing I have realized is that it is my Twilio configuration in pjsip_wizard.conf that was killing me. I have since removed that entire file from /etc/asterisk and I am able to make "from-internal" context calls (i.e., calls that do not leave the VoIP island).

Re: [asterisk-users] SIP URI set 'telephone-context='

I swear I tested it like that and it didn't work. But its working now so thanks guys for your help. On 17 February 2016 at 13:13, Trey Hilyard wrote: > Agree. All you have to do is: > > Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10\;user=phone) > > I am actually surprised

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: >

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > OK. Let me ask this. Is anything else necessary, except choosing TCP as the > preferred protocol on the client, to make TCP w Asterisk work? At the > moment, I have only changed one line in pjsip.conf from my working UDP > setup: > >

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ;

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--- only this

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

OK. I will report with my findings. It appears increasingly likely that I have done something very silly on my side. It is a little perplexing that the EXACT setup (on the same machine) worked for UDP ... On Wed, Feb 17, 2016 at 8:23 AM, Joshua Colp wrote: > Sonny Rajagopalan

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Sorry, I was not being very clear, Joshua, and thanks for your patience with this issue. I had set pjsip set logger on and core set debug 99. See absolutely zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages are not reaching Asterisk, what could be

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sorry, I was not being very clear, Joshua, and thanks for your patience with this issue. I had set pjsip set logger on and core set debug 99. See absolutely zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages are not reaching Asterisk, what could be the issue? I am a little

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Is there a specific place where I can set logger to log incoming TCP segments from L4? $ netstat -tulpn | grep asterisk | grep LISTEN: tcp0 0 0.0.0.0:8088 0.0.0.0:* LISTEN 10313/asterisk tcp0 0

Re: [asterisk-users] SIP URI set 'telephone-context='

Agree. All you have to do is: Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10\;user=phone) I am actually surprised that the dialplan reload would work without it... On Wed, Feb 17, 2016 at 5:51 AM A J Stiles wrote: > On Wednesday 17 Feb 2016, imperium

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Is there a specific place where I can set logger to log incoming TCP segments from L4? $ netstat -tulpn | grep asterisk | grep LISTEN: tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 10313/asterisk tcp0 0 0.0.0.0:50600.0.0.0:*

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: I receive a TCP ack back from that port (5060; owned by Asterisk) --confirmed by wireshark on the Asterisk server. That's from Wireshark, but what is Asterisk seeing? If Asterisk doesn't show the connection or the traffic then something else is up (firewall, etc).

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

I receive a TCP ack back from that port (5060; owned by Asterisk) --confirmed by wireshark on the Asterisk server. What else should I be looking for? This is on a machine on AWS that was running a UDP based Asterisk fine (I did not make ANY other change other than changing protocol=tcp). I also

Re: [asterisk-users] SIP URI set 'telephone-context='

On Wednesday 17 Feb 2016, imperium broadcast wrote: > I kinda have it working with chan_sip. > > Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10;user=phone) > But it doesn't include the user=phone at the end when dialling out. > > "To: ". > >

Re: [asterisk-users] SIP URI set 'telephone-context='

I kinda have it working with chan_sip. Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10;user=phone) But it doesn't include the user=phone at the end when dialling out. "To: ". even adding usereqphone=yes to the sip.conf doesn't add the

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: I can confirm that the server is receiving the SIP request, but simply doesn't do anything with it (log from the server below). Does this have anything to do with how PJSIP was compiled or configured?: TCP support is enabled in PJSIP by default. If you do "pjsip set

[asterisk-users] siemens openstage provisioning

hi, one of my client have hundreds of siemens openstage phones i want implement provisioning (1) for Asterisk and donate the code to some OSS provisioning project can you recommend some "live" provisioning project? thanks (1)

Re: [asterisk-users] 1000 analogue lines with asterisk

On Wednesday 17 Feb 2016, Goke Aruna wrote: > Hello all, > Can someone recommend what hardware to use for a 1000 analogue line > capacity asterisk PABX? > > Regards A PCI express card with four primary rate ISDN ports, each linked up to a channel bank, will give you 120 analogue lines. So you

Re: [asterisk-users] 1000 analogue lines with asterisk

Thanks When I designed the server, I catered for only IP phones but customer want to discard the existing analogue pbx while they want to reuse their cables and analogue phones. Regards On Feb 17, 2016 09:27, "Matt Riddell (lists)" wrote: > There is definitely no way you

Re: [asterisk-users] 1000 analogue lines with asterisk

There is definitely no way you should put 1000 lines on a single box. To be honest I do wonder what you want to do with 1000 lines as your description probably changes the recommendations. Kind regards, Matt > On Feb 17, 2016, at 5:09 PM, Goke Aruna wrote: > > Thanks

Re: [asterisk-users] 1000 analogue lines with asterisk

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on