Re: [asterisk-users] asterisk ip authentication
send me a copy of your sip config also make sure dissallow is before allow. Kind Regards Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/29/2013 1:07 AM, james jan wrote: hi all, i've changedallow=all and restarted service. butstill gives488 Not acceptable here The softswitch sends codec g729. "core show translation" says codec g729 alsa installed. On Sun, Jul 28, 2013 at 10:11 PM, Andrew Colin and...@vsave.co.za wrote: I just find it insecure because if someone does hack they can use any codec. I suppose not very insecure but I like to lock things down as much as possible. On 7/28/2013 9:09 PM, Matt Behrens wrote: On Jul 28, 2013, at 2:59 PM, Andrew Colin and...@vsave.co.za wrote: if you say allow=all it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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 -- _ -- 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] limitation on number of contexts in extensions.conf
but it seems that value of variable defined in external file is not getting
populated during the dialplan execution.
My example:
extract from one external file in /etc/asterisk/abc.conf
PROV=1.2.3.4
[abc]
exten = _1X.,1,Dial(SIP/${PROV}/${EXTEN})
and extensions.conf contains:
[globals]
#include abc.conf
if call is made by the user of abc context, variable ${PROV} is having empty
value. Please suggest where is the problem.
Thanks,
Kamlesh
From: asterisk_l...@earthshod.co.uk
To: asterisk-users@lists.digium.com
Date: Fri, 26 Jul 2013 11:12:28 +0100
Subject: Re: [asterisk-users] limitation on number of contexts in
extensions.conf
On Friday 26 July 2013, Kamlesh Kumar wrote:
Thank you Carlos,
you were right, there was one empty file among all included files which
were causing this problem.
Couple of more queries:
Will system performance be affected if there are 20k dialplan
entries(including all external files and contexts) in extensions.conf?
Not by as much as you think, because the dialplan is compiled into an
intermediate form when Asterisk starts (and again when you execute `dialplan
reload`) -- it doesn't have to parse the whole text file for every call.
Can we define variable in external file, and include that external file in
extensions.conf and then use that variable in dialplan?
Yes (and that's a sensible way of doing it anyway). Just remember, a
variable won't have a value until the include statement which includes the
file
with the line that defines it is parsed.
--
AJS
Answers come *after* questions.
-
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Re: [asterisk-users] asterisk ip authentication
hi Andrew, here is my sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=all On Mon, Jul 29, 2013 at 9:17 AM, Andrew Colin and...@vsave.co.za wrote: send me a copy of your sip config also make sure dissallow is before allow. Kind Regards Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/29/2013 1:07 AM, james jan wrote: hi all, i've changed allow=all and restarted service. but still gives 488 Not acceptable here The softswitch sends codec g729. core show translation says codec g729 alsa installed. On Sun, Jul 28, 2013 at 10:11 PM, Andrew Colin and...@vsave.co.za wrote: I just find it insecure because if someone does hack they can use any codec. I suppose not very insecure but I like to lock things down as much as possible. On 7/28/2013 9:09 PM, Matt Behrens wrote: On Jul 28, 2013, at 2:59 PM, Andrew Colin and...@vsave.co.za and...@vsave.co.za wrote: if you say allow=all it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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 -- _ -- 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 vsave logo 2.jpg-- _ -- 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] Sequence of transfers fail
I have a problem transferring calls multiple times using DTMF sequences (#, *2). The scenario is: Transfereecalls Transferor 1 Transferor 1 transfers to Transferor 2 Transferor 2 transfers to Transfer Target When Transferor 2 enters '#' or '*2', Asterisk no longer reacts and the call remains with Transferor 2. I have tested this with Asterisk 11.2 and 11.5 and there is an entry in the Snom forum which seems to describe the same problem with a reference to Asterisk version 1.8. The transfer problem does not exist when using the REFER/NOTIFY/INVITE way which most SIP phones have on-board. jg -- _ -- 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] RTP from pcap file
You can take the pcap trace using tshark or tcpdump command line linux based tool and open the trace in wireshark. Wireshak is visual tool of tcpdum/tshark(corss platform) and you can listen audio of each call. On Fri, Jul 26, 2013 at 10:17 PM, Gianluca Merlo gianluca.me...@gmail.comwrote: Hello James, Il giorno 26/lug/2013 15:50, James Bensley jwbens...@gmail.com ha scritto: Howdy all, Does anyone know of a niffty CLI tool for Linux that can take a PCAP file that was created on a SIP PBX for example, and then dump the payload of the various RTP streams in there into seperate files so I can listen to them? I can go this graphically with Wireshark, but I'd like to script it for automation. Cheers, James. I personally use rtpbreak http://dallachiesa.com/code/rtpbreak/doc/rtpbreak_en.html For similar tasks Gianluca -- _ -- 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] Sequence of transfers fail
I just got access to an older Asterisk 1.6.2.18 box and found that the multiple transfer problem does not exist here. So with 1.6.2.18 I can transfer as often as I wish using DTMF sequences. jg -- _ -- 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 ip authentication
remove disallow completely you are basically saying do not allow anything then allow anything so remove the disallow part and leave allow Kind Regards Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/29/2013 9:48 AM, james jan wrote: hi Andrew, here is my sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=all On Mon, Jul 29, 2013 at 9:17 AM, Andrew Colin and...@vsave.co.za wrote: send me a copy of your sip config also make sure dissallow is before allow. Kind Regards Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/29/2013 1:07 AM, james jan wrote: hi all, i've changedallow=all and restarted service. butstill gives488 Not acceptable here The softswitch sends codec g729. "core show translation" says codec g729 alsa installed. On Sun, Jul 28, 2013 at 10:11 PM, Andrew Colin and...@vsave.co.za wrote: I just find it insecure because if someone does hack they can use any codec. I suppose not very insecure but I like to lock things down as much as possible. On 7/28/2013 9:09 PM, Matt Behrens wrote: On Jul 28, 2013, at 2:59 PM, Andrew Colin and...@vsave.co.za wrote: if you say allow=all it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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:
[asterisk-users] asterisk 1.4 freezes with queues and iax after virtualization
Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. -- _ -- 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 1.4 freezes with queues and iax after virtualization
On 29/07/13 12:15, samuel wrote: Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. Timing could be an issue. Is dahdi installed? Asterisk 1.4 is old and no longer supported. I would suggest upgrading which would also make the timerfd kernel timing source available if you are running on a recent operating system. See https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- _ -- 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 ip authentication
What is the output of g729 show version? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of james jan Sent: Sunday, July 28, 2013 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk ip authentication hi all, i've changed allow=all and restarted service. but still gives 488 Not acceptable here The softswitch sends codec g729. core show translation says codec g729 alsa installed. On Sun, Jul 28, 2013 at 10:11 PM, Andrew Colin and...@vsave.co.za wrote: I just find it insecure because if someone does hack they can use any codec. I suppose not very insecure but I like to lock things down as much as possible. On 7/28/2013 9:09 PM, Matt Behrens wrote: On Jul 28, 2013, at 2:59 PM, Andrew Colin and...@vsave.co.za mailto:and...@vsave.co.za wrote: if you say allow=all it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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 -- _ -- 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 CPU use
Hello, working in a call center where we set up a structure in asterisk. When my voip reaches 150 calls are with bad quality. We do not transcode codec. What I realized using the top command server (CentOS) processing is too high for the asterisk. But the general processor server is down. Would any limitation of Asterisk to use more hardware resources? tks Eduardo attachment: uso_cpu.PNG-- _ -- 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] Sequence of transfers fail
Well, I forgot to add the t or T option to the dial command, which is required to do transfers with DTMF sequences. jg -- _ -- 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] using E1 PRI lines
Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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 CPU use
On 29/07/13 15:22, Eduardo Leones wrote: Hello, working in a call center where we set up a structure in asterisk. When my voip reaches 150 calls are with bad quality. We do not transcode codec. What I realized using the top command server (CentOS) processing is too high for the asterisk. But the general processor server is down. Would any limitation of Asterisk to use more hardware resources? tks Eduardo I think you need to press '1' in top so that it lists the cpu usage of each core. What version of asterisk are you running? What version of centos? Any dahdi cards installed? -- _ -- 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] using E1 PRI lines
On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). -- _ -- 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] Connected Line presentation in 1.8.x upwards
Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some untrusted trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to cause COLP updates to be ignored for a trunk. I need solutions for SIP, IAX and DAHDI, what options do I have? This applies to both in- and out-bound calls. Are there some variables that I can set just before dialling an outbound call, and immediately on receiving an inbound call to determine what the callerID values will be for the entire duration of the call? (ie. old-style pre-COLP behaviour for specific trunks) Thanks for any pointers. Regards, Steve -- _ -- 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] Connected Line presentation in 1.8.x upwards
From: Steve Davies davies...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 07/29/2013 10:53 AM Subject:[asterisk-users] Connected Line presentation in 1.8.x upwards Sent by:asterisk-users-boun...@lists.digium.com Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some untrusted trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to cause COLP updates to be ignored for a trunk. I need solutions for SIP, IAX and DAHDI, what options do I have? This applies to both in- and out-bound calls. Are there some variables that I can set just before dialling an outbound call, and immediately on receiving an inbound call to determine what the callerID values will be for the entire duration of the call? (ie. old-style pre-COLP behaviour for specific trunks) Thanks for any pointers. Regards, Steve I believe what you are looking for in Dial is the 'I' option. -- _ -- 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] using E1 PRI lines
On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). I am using digium card TE410P. can anyone help me how to change jumper settings -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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 1.4 freezes with queues and iax after virtualization
there's no dahdi installed. Following debugging the issue, it looks like the astdb file is broken. Whenever database show command is executed it loops over the same results. The file itself is around 225K but dumping its content via asterisk -rx 'database show' creates and endless file. Is there any easy way to restore the database content? Thanks a lot for the replies, Samuel. On 29 July 2013 14:07, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 29/07/13 12:15, samuel wrote: Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. Timing could be an issue. Is dahdi installed? Asterisk 1.4 is old and no longer supported. I would suggest upgrading which would also make the timerfd kernel timing source available if you are running on a recent operating system. See https://wiki.asterisk.org/** wiki/display/AST/Timing+**Interfaceshttps://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- __**__**_ -- 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-**usershttp://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] using E1 PRI lines
hello:you can add T1_E1 by load card drivers Best regards, James.zhu website: www.hiastar.com From: akibsay...@gmail.com Date: Mon, 29 Jul 2013 21:48:19 +0530 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] using E1 PRI lines On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). I am using digium card TE410P. can anyone help me how to change jumper settings -- _ -- 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 -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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] using E1 PRI lines
I didnt understand what you were saying.can you please explain I am using digium cards sent from android On Jul 30, 2013 6:55 AM, James zhu zhulizh...@live.com wrote: hello: you can add T1_E1 by load card drivers Best regards, James.zhu website: www.hiastar.com -- From: akibsay...@gmail.com Date: Mon, 29 Jul 2013 21:48:19 +0530 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] using E1 PRI lines On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). I am using digium card TE410P. can anyone help me how to change jumper settings -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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 -- _ -- 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] using E1 PRI lines
On 30/07/2013, at 4:22 PM, Akib Sayyed akibsay...@gmail.com wrote: I didnt understand what you were saying.can you please explain I am using digium cards sent from android E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC connectors ) or a 120 ohm balanced twisted pair. The other standard is T1 and digium cards can let you choose between T1 E1 and definitely do 120 ohm Telco's will usually provide 120ohm twisted pair interfaces as it travels further and has less interference from noise. On Jul 30, 2013 6:55 AM, James zhu zhulizh...@live.com wrote: hello: you can add T1_E1 by load card drivers Best regards, James.zhu website: www.hiastar.com From: akibsay...@gmail.com Date: Mon, 29 Jul 2013 21:48:19 +0530 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] using E1 PRI lines On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). I am using digium card TE410P. can anyone help me how to change jumper settings -- _ -- 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 -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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 -- _ -- 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] using E1 PRI lines
Operators are unnecessarily confusing you by talking tech Lang which you
are not well versed with. Are you trying to create prod / services which
they don't want u to launch but they have to provide lines under some sort
of regulatory obligations ?
Just go ahead n plug the wires on the E1 card ports.
Mitul
On Tuesday, July 30, 2013, Duncan Turnbull wrote:
On 30/07/2013, at 4:22 PM, Akib Sayyed
akibsay...@gmail.comjavascript:_e({}, 'cvml', 'akibsay...@gmail.com');
wrote:
I didnt understand what you were saying.can you please explain
I am using digium cards
sent from android
E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC
connectors ) or a 120 ohm balanced twisted pair.
The other standard is T1 and digium cards can let you choose between T1
E1 and definitely do 120 ohm
Telco's will usually provide 120ohm twisted pair interfaces as it travels
further and has less interference from noise.
On Jul 30, 2013 6:55 AM, James zhu zhulizh...@live.comjavascript:_e({},
'cvml', 'zhulizh...@live.com');
wrote:
hello:
you can add T1_E1 by load card drivers
Best regards,
James.zhu
website: www.hiastar.com
--
From: akibsay...@gmail.com javascript:_e({}, 'cvml',
'akibsay...@gmail.com');
Date: Mon, 29 Jul 2013 21:48:19 +0530
To: asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
'asterisk-users@lists.digium.com');
Subject: Re: [asterisk-users] using E1 PRI lines
On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk javascript:_e({}, 'cvml',
'mailinglist+aster...@dns99.co.uk'); wrote:
On 29/07/13 16:28, Akib Sayyed wrote:
Dear asterisk users
I wanted to use E1 pri lines on my asterisk box but my provider support
only 120ohm on E1 line. I dont know how to set those values.
Please help me
Its done on whatever interface cards you have. Some may have a jumper
setting. I know Sangoma has it in their configuration file (wanpipe).
I am using digium card TE410P. can anyone help me how to change jumper
settings
--
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Akib Sayyed
Matrix-Shell
akibsay...@gmail.com javascript:_e({}, 'cvml', 'akibsay...@gmail.com');
akibsay...@matrixshell.com javascript:_e({}, 'cvml',
'akibsay...@matrixshell.com');
Mob:- +91-966-514-2243
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Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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