Re: [asterisk-ss7] CRBT with ss7 and asterisk

Hi , You are looking only CRBT function or total calls route via asterisk server . On Thu, Nov 10, 2011 at 4:29 PM, Peter Hyeroba wrote: > Hello All, > > Has anyone implemented CRBT with ss7 on asterisk, > > what hardware would one recommend for over 100,000 concurrent calls. > > I

Re: [asterisk-ss7] CIC numbering in chan_ss7

Hi , Try This [channels] ;rxwink=300 language=en context=apex callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes networkindicator=national ss7_called_nai=national ss7_calling_nai=nationa

Re: [asterisk-ss7] libss7 terminate digit

or my poor english > > Regards > German > > *From:* bipin singh > *Sent:* Wednesday, November 02, 2011 12:32 AM > *To:* asterisk-ss7@lists.digium.com > *Subject:* Re: [asterisk-ss7] CIC isn't same of Telco > > Hi, > In this case sigchan=16 on both ss7

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 13

Hi, First check you maximum concurrent calls happen on you server . CRBT is maximum 30 to 45 sec process so approx you process two calls per mint per channels . On Fri, Nov 11, 2011 at 12:52 PM, Peter Hyeroba wrote: > The flow is a call comes in from the pstn and the switch forwards

Re: [asterisk-ss7] Some Channels are Blocked with remote maintanence

Hi , Try asterisk-1.6.X . On Mon, Nov 21, 2011 at 2:05 PM, Stefan Schmidt wrote: > Hello list, > > i have two E1 running on a asterisk 1.8 with chan_ss7 which i have set > to production state last week. And after around 5000 calls i can see > that 3 channels are in state BLOCKED R

Re: [asterisk-ss7] two DPC with same cic numbering

Hi, check your conf how many time configured linkset=2 . Currect it first then check output . On Fri, Nov 25, 2011 at 8:43 PM, Cees Harteveld < cees.hartev...@budgetphone.nl> wrote: > Hello, > > > Maybe some one knows the answer, > > ** ** > > ** ** > > I got two linkset set up w

Re: [asterisk-ss7] Using Asterisk for SS7 Signalling only

Hi, Check it http://www.voip-info.org/wiki/index.php?page_id=2714 http://wiki.sangoma.com/wanpipe-linux-asterisk-ss7 On Mon, Nov 28, 2011 at 6:36 PM, Abdul Hakeem wrote: > Hello, > > Could someone explain the process on how to use an Asterisk for SS7/ISUP > signalling with Cisco AS5350

Re: [asterisk-ss7] Using Asterisk for SS7 Signalling only:10

It's Work? 2011/12/1 Horacio J. Peña > I assume Krzysztof was talking about my converter. > > You can get it on http://200.69.223.160/~horape/H7.tgz > > It's not ready for public release (and unless somebody else adopts it, it > will > probably never be) but it's been working good for a year or

Re: [asterisk-ss7] Using Asterisk for SS7 Signalling only:10

Hi, SS7 to PRI or else On Fri, Dec 2, 2011 at 6:39 PM, Krzysztof Drewicz < krzysztofdrew...@gmail.com> wrote: > > > 2011/12/1 Horacio J. Peña > >> I assume Krzysztof was talking about my converter. >> >> > Yes Horacio. > > > >> >> >> There are no >> docs but you should be able to under

Re: [asterisk-ss7] Digium PRI card with WARNING[8097]: l4isup.c:5069 l4isup_event: Received CIC=1 for unequipped circuit (typ=GRS), link 'l1'.

Hi,[channels] ;rxwink=300 language=en context=apex callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes networkindicator=national ss7_called_nai=national ss7_calling_nai=national ss7_explictacm=yes ss7_in

Re: [asterisk-ss7] SS7 + T1 on Asterisk?

Hi, It is possible on both T1 and E1. On Wed, Dec 7, 2011 at 10:17 PM, Matt wrote: > If I were to get a 2 span T1 card for Asterisk... and connect it to a > Nortel DMS100... can I run call traffic over the T1 and run SS7 > signaling FOR the T1 over the other port? > > Is there docum

Re: [asterisk-ss7] first time SS7 installation

Hi , Use asterisk-1.6.X and dahdi with libss7 . On Wed, Dec 7, 2011 at 6:10 PM, Vieri wrote: > Hi, > > I'm trying to configure SS7 for the first time on a system that I cannot > currently upgrade (ie. I need to use zaptel and Asterisk 1.4): > > asterisk 1.4.31 > zaptel 1.4.12.1 > libpri

Re: [asterisk-ss7] Unable to forward frametype: 2

Hi, May be CIC setting issue please check your proper ACP and DCP connection. First try only single link. It may be help you . Find the chan_dahdi.conf of working on Ericsson and Siemens switch . [channels] ;rxwink=300 language=en context=apex callwaiting=yes callwaitingcallerid=yes thre

Re: [asterisk-ss7] Asterisk 1.4.29.1 // chan_ss7-1.4.3 // Digium TE420P Signalling lost

Hi , Try libss7 and asterisk-1.6.X . On 1/12/12, Diego Reinoso wrote: > Few days ago I lost signalling in my asterisk box. > > I am unaviable to capture signalling packets from "ss7 dump", my card its > ok, "ss7 dump" creates the fil but it is empty, I execute "ss7 reset" but > noth

Re: [asterisk-ss7] Excessive poll issue with chanss7

Hi , Try libss7 and asterisk-1.6.X may be get you some positive response . On Wed, Jan 18, 2012 at 11:25 PM, Vallimamod ABDULLAH < vallimamod.abdul...@imtelecom.fr> wrote: > Hi Robert, > > On Jan 18, 2012, at 5:34 PM, Robert Thomas wrote: > > > I spoke with Anders about this probl

Re: [asterisk-ss7] Presentation in Original Called Number and Redirecting Number

Hi, It's only possible when you directly connected with switch node. On Wed, Jan 18, 2012 at 9:16 PM, peterpet wrote: > Hello, > > is it possible to set in a Call Forwarding scenario the Presentation to 0 > (presentation=0 (allowed)) in the Original Called Number and in the > Redirect

Re: [asterisk-ss7] Mantis ID 13495 - READY TO TEST

Hi, Provide the bug log ... On Fri, Jan 20, 2012 at 4:32 PM, Kaloyan Kovachev wrote: > On Mon, 16 Jan 2012 11:05:04 +0200, Kaloyan Kovachev > wrote: > >>> On 01/04/2012 07:49 AM, Kaloyan Kovachev wrote: > Hello list, > most of us are using Domjan's version with Aster

Re: [asterisk-ss7] chan_ss7 incoming callbycall call problems

Hi , If possible try libss7. On Sat, Mar 3, 2012 at 2:06 AM, Simon Handschin wrote: > Hi thank you for your fast answer. > Its about carrier pre selection eg i make a call from an external phone > (not connected to the server) with our cpc like 1234. > > 12340041441234567 > > The Call

Re: [asterisk-ss7] For help

Hi, Yes, Try without CIC setting and also block from provider end. On Thu, Mar 1, 2012 at 12:52 PM, Bai Bin wrote: > hello,everyone: > > i'm a freshman on voip. there is a problem about asterisk . > > there is a 4E1 with signalling(ss7) and three servers(a part has one > s

Re: [asterisk-ss7] For help

020202 > dpc => g1:0x010101 > links => l1:1,l2:2,l3:3,l4:4,l5:5,l6:6,l7:7,l8:8,l9:9,l10:10,l11:11,l12:12 > if-1 => 192.168.2.92 > > [host-host2] > enabled => yes > default_linkset => g1 > opc => 0x020202 > dpc => g1:0x010101 > links => > l13:1,l14:2,l

Re: [asterisk-ss7] Error my_handle_link_exception: SS7 got event: HDLC Abort(6) on span

Hi , Try any old version of dahdi driver . 2012/3/13 Gustavo Mársico > I saw these messages when the STM-1 (or converter or any kind of box that > connects the E1 to the digium card) is loosing sync with the far end. > > > On Mar 12, 2012, at 11:04 PM, James zhu wrote: > > > hel

Re: [asterisk-ss7] different context on each spam

Hi , First map to all campaign to different different group and then set simple context=your context name example group=1 cicbeginswith = 1 sigchan = 16 channel=1-15 cicbeginswith = 17 channel = 17-31 context=your context name group=2 cicbeginswith = 33 channel = 32-46 ;sigchan = 47

Re: [asterisk-ss7] no audio with many groups

Hi , Its not group issue check your other things. On Sat, Mar 17, 2012 at 1:03 AM, Marcus Vinicius wrote: > Hi, > > I have an linkset like this: > > linkset=1 > pointcode=1 > defaultdpc=2 > adjpointcode=2 > > context=from-pstn-01 > group=1 > cicbeginswith=1 > channel=>1-15 > cicbegi

Re: [asterisk-ss7] Variable if call is national or not

Hi, Try this networkindicator=national ss7_called_nai=national ss7_calling_nai=national On Thu, Apr 5, 2012 at 10:41 PM, Torrey Searle wrote: > I see no reason dynamic should not work for you. N will be added to > anything received with a national cli. but you will need to be sure > to prefix

Re: [asterisk-ss7] REL sent to the carrier

Hi , It may be some configuration issue? check some another asterisk and dahdi version. On Sat, Apr 21, 2012 at 4:59 AM, her Garcia wrote: > Hi. > > I am having an issue with REL messages. My carrier tells me that he > receives my calls starting with an IAM and 2 seconds later a REL/Hang up > me

Re: [asterisk-ss7] Call conected but not ringing

Hi, Try it without sip user and single ss7 link. If possible try some incomings calls if link is ok then calls land on server . On Fri, Apr 20, 2012 at 6:45 PM, her Garcia wrote: > Hi, everyone. I am working on asterisk+ss7. > When I try to make a call, the call connects but I have no audio or

Re: [asterisk-ss7] libss7 / Asterisk 10 / t.38

Hi, without T.38 gateway. On Mon, Apr 23, 2012 at 8:02 PM, Kevin P. Fleming wrote: > On 04/22/2012 10:17 AM, Gregory Massel wrote: > >> Hello >> >> Is anyone using libss7 on Asterisk 10 with T.38 gateway (or without >> using T.38)? >> >> I've got two boxes on Asterisk 1.8 with libss7 (in addition

Re: [asterisk-ss7] dahdi, libss7 and asterisk

Hi, Try only SS7 links other asterisk box. On Sun, Apr 29, 2012 at 1:28 AM, Vishnu Kumar D R wrote: > Hi, > > Heres my whole chan_dahdi.conf > > [channels] > #include /etc/asterisk/dahdi-channels.conf > usecallerid=yes > callwaiting=yes > restrictcid=yes > usecallingpres=yes > callwaitingcallerid

Re: [asterisk-ss7] dahdi, libss7 and asterisk

: > Try moving your sigchan line before your channel= declaration > > I think in the chan_dahdi code the linkset is only actually created > the moment it hits the first sigchan= line > > Torrey > > On 4 May 2012 06:11, bipin singh wrote: > > Hi, > > T

Re: [asterisk-ss7] chan_ss7 T35 Timer does not work properly

Hi , Check your Aastra phone PBX setting. On Mon, May 14, 2012 at 1:56 PM, Simon Handschin wrote: > Hello everybody > > I have a system with Asterisk Asterisk 1.8.9.3 and chan_ss7-2.1.0. > > If i make a Overlap Dial from a normal analog phone the the dial goes > into the dialplan after the 4000 m

Re: [asterisk-ss7] Asterisk trap divide error error:0 in res_musiconhold.so

Hi, Don't use your own mp3 files in music on hold after that check it. On Tue, May 8, 2012 at 2:22 PM, Hanie Maghsoudy wrote: > Hello all, > > My asterisk service crashes time to time and I receive this error in > /var/log/message: > > May 8 08:59:24 MyServer kernel: asterisk[3805] trap divide e

Re: [asterisk-ss7] ALIGNMENT Problem

Hi, Try This [channels] ;rxwink=300 language=en context=apex callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes networkindicator=national ss7_called_nai=national ss7_calling_nai=national ss7_explictacm=

Re: [asterisk-ss7] Looking for advice in CIC numbering for E1 upgrade

Hi , If you have single point code (adjpointcode) then it's ok . Also you try single sigchan for all 8 port. On Fri, Jun 1, 2012 at 6:48 AM, Luis Marcelo wrote: > Hey guys, > I wanted to get some advice for an upgrade we have in mind: > Scenario: We have 1 server with 2xA104d Sangoma cards (8E1s

Re: [asterisk-ss7] F on the end of the number

Hi, Check your dialplan setting both side(switch and asterisk). On Mon, Jul 30, 2012 at 10:49 PM, Marcelo Pacheco wrote: > That's the beauty of free software. > > The layout I defined suits my needs. > > Ok, so it can't be the default for most others. > > I never send/receive #/* through ISUP bec

Re: [asterisk-ss7] Asterisk + libss7 with BT in the UK

Yes, Direct BT connected and working . On Wed, Oct 17, 2012 at 2:11 PM, Torrey Searle wrote: > You should try looking at the libss7 branch from atilla, it exposes > alot more timers > > alternatively, you can try one of the patches from > > https://issues.asterisk.org/jira/browse/SS7-27 > > to g

Re: [asterisk-ss7] Asterisk ISUP to SIP mapping

Yes, It's possible 100%. Asterisk use codec translator for that . On Mon, Oct 8, 2012 at 3:37 PM, Charl Barnard wrote: > Hi, > > Indeed you can-I've done it many times, and it works well. Asterisk can > even do the media translation for you as well, so for example the G.711 > alaw used with SS7

Re: [asterisk-ss7] LibSS7 and Sangoma A101 - SS7 HDLC Overrun Error

Hi, load/interrupt related issue may be. Check it on single card and single spam . On Fri, Oct 12, 2012 at 3:20 AM, Gregory Massel wrote: > This does not appear to be an SS7 issue. > > Check your cables and connections, particularly for loose contacts or > potential shorts (e.g. where the jumper

Re: [asterisk-ss7] Zombie Channels

Hi, Check your php AGI properly , also check it without dataabse and AGI, may be your database creating issue. On Thu, Nov 15, 2012 at 5:51 PM, [Digital^Dude] ® wrote: > Hello, > > I am using asterisk 1.6.x and 1.8.18.0 (LTS) on a CentOS 5 boxes. I'm > using php AGI for incoming calls and after

Re: [asterisk-ss7] charge indicator

Hi , Check it http://www.ss7-training.net/sigtran-training/ch09lev1sec4.html On Fri, Jan 18, 2013 at 5:14 PM, Laura Maurizi wrote: > Hi Marcelo, > > I agree to pay for your work and, since you have experience in SS7 I’ll > explain to you what else I need. > > I’m trying to interconnect

Re: [asterisk-ss7] No audio in ss7 link

Hi pool, Check incoming IVR calls and also DTMF input., On Mon, Feb 18, 2013 at 4:51 PM, rubel pool wrote: > Dear all, > > In my ss7 link, my link is up with all configuration My connectivity > with carrier is 2 E1.. One E1 for signalling and another E1 for Voice.. > > so from last few year

Re: [asterisk-ss7] No audio with CON message

Hi try without SIP user or IVR calls. On Thu, Feb 14, 2013 at 7:48 PM, Attila Domjan wrote: > Just curious. Moved all of our SS7 interconnects to SIP. > > On Thu, 2013-02-14 at 16:15 +0200, Kaloyan Kovachev wrote: > > I am so glad to see you back Domjan > > > > On Thu, 14 Feb 2013 14:39:37 +0100,

Re: [asterisk-ss7] IAM received as Warning

Hi , It's CIC channel mapping issue from Telecom. On Fri, Feb 8, 2013 at 2:51 PM, Salman Minhas wrote: > Hi, > > Anyone using Chan_ss7_2.1.0_svn ? IAM receiving from Telco shown as a > Warning in asterisk. > > *WARNING*: l4isup.c l4isup_event: Received IAM (CIC 223), link 'l5'. > > *WARNING*: l4

Re: [asterisk-ss7] chan_ss7 | Link established but no voice

Hi, Check first any ivr calls via ss7 link. After that try to sip to sip. If all case are ok then go for your setting. may be sip codec issue. On Wed, Mar 20, 2013 at 12:01 PM, Vinod Dharashive wrote: > Hi , > > You can download SVN trunk as below > > svn co http://svn.netfors.com/chan_ss7/trunk

Re: [asterisk-ss7] CDR Data

Hi , share the chan_dahdi.conf and dialplan On Sun, Apr 21, 2013 at 7:01 PM, Nyamul Hassan wrote: > Thank you Atilla! > > On Sun, Apr 21, 2013 at 7:13 PM, Attila Domjan > wrote: > >> In my system are used lot of custom cdr variables for many similar >> cases... >> > > Yeah, I was thinking of

Re: [asterisk-ss7] getting MTP receivebuf issue (Urgent Help)

Hi, Check the h/W IRQ and dahdi_test: output. On Sun, May 26, 2013 at 4:11 AM, Muhammad Shahzad < engrmuhammadshah...@hotmail.com> wrote: > Dear All, > > ** ** > > This is my first post on this mailing list, So my hopes are so high. > > ** ** > > I am getting MTP receivebuf issue, The pr

Re: [asterisk-ss7] moh stopped before ringing

Hi, Check your default music on hold files and setting musiconhold.conf for music on hold . First check it SIP to SIP after that try it on SS7. otherwise remove the music on hold from dialplan extensions.conf. On Sun, Jun 2, 2013 at 4:55 PM, 胡万勇 wrote: > my environment: > centos 5.4 > asteris

Re: [asterisk-ss7] moh stopped after ringing

Hi, Check your default music on hold files and setting musiconhold.conf for music on hold . First check it SIP to SIP after that try it on SS7. otherwise remove the music on hold from dialplan extensions.conf. On Sun, Jun 2, 2013 at 4:37 PM, 胡万勇 wrote: > my environment: > centos 5.4 > asteris

Re: [asterisk-ss7] Asterisk + chan_ss7 2.2.0 no voice problem

Hi, Try libss7 for ss7 signalling . http://downloads.asterisk.org/pub/telephony/libss7/ On Thu, Jun 20, 2013 at 2:34 PM, James zhu wrote: > hello: > Run: ss7 link status under Asterisk CLI, and check with service the first > cic. > > Best regards, > James.zhu > Connect with Sangoma

Re: [asterisk-ss7] how does APC work in libss7?

Hi , Set APC and DCP same. check the configuration : linkset=3 pointcode=504 adjpointcode=9380 defaultdpc=9380 ss7_called_nai=national ss7_calling_nai=national networkindicator=national sigchan=78 cicbeginswith=1 channel=63-77 cicbeginswith=17 channel=79-93 On Wed, Jun 19, 2013 at 8:32 PM, Marc

Re: [asterisk-ss7] Echo and Noise problems with chan_ss7

Hi , Try libss7 for ss7 signalling. On Wed, Jun 12, 2013 at 1:03 PM, Simon Handschin wrote: > Hi Everybody! > > I have some problems with echo and noises during calls. I played a bit > with echo_train and echo_taps but no change. > > Have someone similar problems? > > > Here is my setup: > > Cha

Re: [asterisk-ss7] Echo and Noise problems with chan_ss7

ct With Sangoma > www.hiastar.com > > - Original Message ----- > *From:* bipin singh > *To:* asterisk-ss7@lists.digium.com > *Sent:* Friday, June 21, 2013 2:04 PM > *Subject:* Re: [asterisk-ss7] Echo and Noise problems with chan_ss7 > > Hi , > Try libss7 for ss7 sign

Re: [asterisk-ss7] No Voice in channels

Hi, First Check Single ss7 link with single ACP and DCP. On Fri, Aug 30, 2013 at 6:48 PM, Akib Sayyed wrote: > I did configurations but still there is no voice. > > > On Thu, Aug 29, 2013 at 6:21 PM, Frederic Van Espen > wrote: > >> On 08/29/2013 01:55 PM, Akib Sayyed wrote: >> >>> cicbeginswi

Re: [asterisk-ss7] redirecting number on outgoing calls

Hi , Same thing confirm from provider. Also check the RDNIS variable. On Tue, Aug 20, 2013 at 6:55 PM, Kaloyan Kovachev wrote: > Hi, > > > On 2013-08-20 15:21, Frederic Van Espen wrote: > >> Hi, >> >> There's contradicting information available online about redirecting >> number and I'd like to

Re: [asterisk-ss7] Megaco

Hi, Currently asterisk not supported . On Tue, Aug 20, 2013 at 8:16 AM, Bruno Rodrigues wrote: > Laura, > > As far I know Asterisk have suport to MGCP only as Controler not as > Gateway, what are the softswitchs needs to controler the Asterisk Box, will > the softswitch control only the voice or

Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

Its supported SIP protocol or else. On Fri, Aug 16, 2013 at 9:07 PM, Nick Khamis wrote: > Pay 8,000 USD for the Audiocodes SS7 license and forget about > asterisk-ss7. > > -- > _ > -- Bandwidth and Colocation Provided by ht

Re: [asterisk-ss7] SS7-ISUP

echo in IVR or echo in calls check it both are different ? On Thu, Oct 10, 2013 at 1:32 AM, Marcelo Telco wrote: > > Dear friends mailing list, > > I have a problem using SS7 with a telecommunications carrier. The E1 trunk > is setting, making calls, but he did not receive. It shows the message

Re: [asterisk-ss7] How to install Sangoma A116 with chan_ss7 and asterisk

If possible try libss7 for ss7 signalling . On Tue, Oct 8, 2013 at 7:12 AM, James zhu wrote: > hello, Guys: > there is a doc for how to install Sangoma A116, chan_ss7 and asterisk. I > use STP code > with first channel for signalling. It is very unique and good example for > how to install that

Re: [asterisk-ss7] SS7-ISUP

Hi, Check it with IVR also record the same calls with in out format. On Thu, Oct 24, 2013 at 2:39 AM, Camilo Echeverry wrote: > Have you configured some destination for incoming calls in your dialplan? > could be something like this: > > you may change SIP/1000 for the technology/extension you

Re: [asterisk-ss7] No audio using libss7 over E1 (SOLVED)

On Fri, Aug 3, 2012 at 1:06 AM, Krzysztof Drewicz < krzysztofdrew...@gmail.com> wrote: > > > 2012/8/2 Marcelo Pacheco > >> No, THATS A BAD SS7 EXPERIENCE. >> You must learn this BEFORE trying to mess around with ISUP. >> > > > And the very 1st posts suggested - to pull cable out. > But, nowadays

Re: [asterisk-ss7] Looking for advice in CIC numbering for E1 upgrade

On Fri, Jun 1, 2012 at 9:45 AM, bipin singh wrote: > Hi , > If you have single point code (adjpointcode) then it's ok . Also you try > single sigchan for all 8 port. > > On Fri, Jun 1, 2012 at 6:48 AM, Luis Marcelo wrote: > >> Hey guys, >> I wanted to get so

Re: [asterisk-ss7] different context on each spam

On Wed, Mar 14, 2012 at 8:45 AM, bipin singh wrote: > Hi , > First map to all campaign to different different group and > then set simple context=your context name > example > group=1 > > cicbeginswith = 1 > sigchan = 16 > channel=1-15 > cicbeg

Re: [asterisk-ss7] For help

On Wed, Mar 7, 2012 at 10:29 AM, bipin singh wrote: > Hi , >Check it 2 ss7 links configuration but it is on libss7. In > this configuration both ss7 link working on single ACP and DCP. > It is working on Ericsson switch . > > [trunkgroups] >

Re: [asterisk-ss7] Redundant sigchan, libss7

On Tue, Mar 6, 2012 at 7:29 PM, Kaloyan Kovachev wrote: > On Tue, 6 Mar 2012 05:42:24 -0800 (PST), Marcus Vinicius > wrote: > > Hi > > > > dahdi status: > > > > > > *CLI> dahdi show status > > Description Alarms IRQbpviol CRC4 > Fra > > Codi Options LBO > > AP40

Re: [asterisk-ss7] DTMF Issue

On Sat, Mar 10, 2012 at 1:05 AM, Ryan Crowder wrote: > That did it! Thank you Kevin for the fast responses. > > -Original Message- > From: asterisk-ss7-boun...@lists.digium.com > [mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Kevin P. > Fleming > Sent: Friday, March 09, 2012

Re: [asterisk-ss7] For help

On Thu, Oct 24, 2013 at 10:08 AM, bipin singh wrote: > > > > On Wed, Mar 7, 2012 at 10:29 AM, bipin singh > wrote: > >> Hi , >>Check it 2 ss7 links configuration but it is on libss7. In >> this configuration both ss7 link working on single ACP

Re: [asterisk-ss7] ss7 Link reset after 30 seconds

Hi, Check your CRC4 setting On Wed, Dec 4, 2013 at 7:50 PM, peterpet wrote: > Hi, > > everithing is good. But every 30 seconds circuits are in reset. > > This is from ss7 debug: > > [Dec 4 16:11:02] WARNING[7511]: chan_dahdi.c:10094 ss7_linkset: MTP2 link > down (SLC -1214249800) > Changeover

Re: [asterisk-ss7] Received message destined for point code 0x762 but we're 0x154a. Dropping

Check it on span 1 . It may be CIC issue closed all other CIC from telco side if you are usinf more than one ss7 link. On Tue, Dec 24, 2013 at 9:54 AM, Huan Nguyen Hoang < nguyenhoanghua...@gmail.com> wrote: > I think so. > > > 2013/12/23 Vahan Yerkanian > >> On Dec 22, 2013, at 10:17 PM, peter

Re: [asterisk-ss7] Looking for help ss7 signal sharing with multiple servers

If possible take two dchan for same DCP. IT support desk: i...@essencekey.com www.essencekey.com On Mon, Mar 3, 2014 at 8:23 AM, Network Solution Bagladesh < networksolutio...@gmail.com> wrote: > Hi All, > Thanks for your reply. All E1s are working fine if we use a single server. > But main goa

Re: [asterisk-ss7] Looking for help ss7 signal sharing with multiple servers

> > Thanks in Advance > > > On Mon, Mar 3, 2014 at 9:42 AM, bipin singh wrote: > >> If possible take two dchan for same DCP. >> >> IT support desk: >> i...@essencekey.com >> www.essencekey.com >> >> >> On Mon, Mar 3, 2014 at 8:23 AM, N

Re: [asterisk-ss7] chan_ss7 or libss7 work with asterisk v13?

Hi, Every E1 have only 30 voice channels so you need 30 CIC for them. Find the detail of two E1 in same linkset sigchan=16 cicbeginswith=1 channel=1-15 cicbeginswith=17 channel=17-31 ;;2 E1 on same linkset ;;; sigchan=47 cicbeginswith=33 channel=32-46

Re: [asterisk-ss7] Two servers with same point code - chan_ss7

You can use same point code from telco side but your end need different point code in libss7. On Fri, Jul 15, 2016 at 5:44 AM, Rafael dos Santos Saraiva < rafaels...@gmail.com> wrote: > My scenario: https://s32.postimg.org/jybtpsdr9/same_opc.png > > > 2016-07-14 5:37 GMT-03:00 Mesbahuddin Malik

Re: [asterisk-ss7] Force out-bond call to specific CIC

same CIC mapping required to telco side try to incoming call on same CIC . May be some telco end mapping issue . On Thu, Jul 14, 2016 at 6:23 AM, Rafael Prado Rocchi wrote: > Hi, > I'm considering you are using libss7 implementation, you can access CIC > directly using the same syntax dahdi uses

Re: [asterisk-ss7] SS7 Linkset issue...

Hi, CIC not to map channels ,first map to channels and check ss7 link status. On Wed, Dec 14, 2016 at 5:57 PM, Mitul Limbani wrote: > Hi Andre, > > Looks like your CICs are not aligned physically. > > How many E1s are you configuring and what is the Signalling CIC and which > E1 is carrying thi

Re: [asterisk-ss7] Two stp in chan_ss7

i think , Its not possible On Thu, Jun 22, 2017 at 9:55 PM, Rafael dos Santos Saraiva < rafaels...@gmail.com> wrote: > Hi > > Is it possible using two or more stp in a link using chan_ss7? > > I tried two ways of this: > > ;comma separated > stp => 123,456 > > ;one per line > stp => 123 > stp =>

Re: [asterisk-ss7] SS7 in asterisk 1.4 ?

Hi u use asterisk-1.6 or asterisk-trung u got all cli in asterisk and its work fine bipin raghuvanshi Sr. software eng. INDIA On 12/24/08, Ruddy Gbaguidi wrote: > Hi All > I have read a lot to be able to setup our upcoming SS7 link from our Telco. > I finally choose to go with libss7 since it

Re: [asterisk-ss7] SS7 Config

hi first you set your pointcode and dpc currect . this is main problem on link u got the signalling.. On 1/12/09, Ruddy Gbaguidi wrote: > > Hi All > I have to setup an ss7 link with the telco but they are not seeing any > signal from my side. > > Behind the digium card the light

Re: [asterisk-ss7] Need some information on SS7 parameters

Hi i dont understand vodaphone send BSE OR MSI id both case incoming and outgoing line. i m useing vodaphone 2 ss7 link. On 2/3/09, resea...@businesstz.com wrote: > > Hello List > > I am setting up a small demo site using SS7 and one of the requirement is > to be able to unhide the numbers and l

Re: [asterisk-ss7] chan_ss7 one way adio

no nothing problem check ur voice server On 2/24/09, Mathias Doerr wrote: > > Hello, > I`m using chan_ss7-1.0.10 with Digium TE410 card . Very often is one way > audio where the called party can`t hear the callee, I `ve seen this problem > was also reported by others, is there any fix available

Re: [asterisk-ss7] do i need to install libpri

no need libpri only use libss7 www.ehorizons.in On Sat, Apr 18, 2009 at 4:56 PM, Kashif Ali wrote: > Dear all, > > i wan to configure digium te412p card as ss7 interface. i installed > libss7,dahdi and asterisk. Do i need to install lip pri also to work on ss7? > > best regards. > Kashif > >

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

hardware irq missing try another server or use lspci -bv On 4/18/09, Gobinda Paul wrote: > > Hi, > > I am facing some problem to configure ss7 with asterisk.Here is my > configuration bellow. > > I am using digium 410p card, According to my telco, > Signal Chanel: 1st Channel Of First E1, > My p

Re: [asterisk-ss7] Newbie - Asterisk + PRI without SS7 support

India telecos support SS7 signalling. not clear location information only MSC id . On 4/22/09, technologyinspired wrote: > > Hi, > > I am new when it comes to SS7. I am using Asterisk now since last 1 year > and have written complex IVRs and done deployments. My question is ve

Re: [asterisk-ss7] SS7 linkset down.

Hi Rana use sigchan=16 up to configuration on channels its work try. On Sun, May 3, 2009 at 9:26 PM, Rana Dhekial wrote: > Hi , > > I am having problem with the first span of the Digium TE420 card. "dahdi > show status" was showing span 1 as red. The SS7 linkset was up when I us

Re: [asterisk-ss7] SS7 linkset down--->working

Hi check its work sigchan=16 > move this to line above channel => 1-15 because i m using this configuration in multi setup and its work On Tue, May 5, 2009 at 11:12 PM, Rana Dhekial wrote: > Following config seems to be working .. > > > > /etc/dahdi/system.c

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

Hi ur ss7 links not working properly .reset ur link On Thu, May 14, 2009 at 5:58 AM, Gobinda Paul wrote: > Hi, > > Any Idia? > > *if i dial any number , i got 34 error. can you please tell what's the > wrong done by me? > * > -- Executing [8801720039...@default:1] > Answer("SIP/X

Re: [asterisk-ss7] HOWTO Asterisk + LibSS7

Hi use asterisk-1.6 latest dahdi latest version and libss7 . 1. dahdi kernel install 2. dahdi tool install 3. libss7 4. asterisk On Fri, Jul 3, 2009 at 2:02 PM, Saeed Ahmed wrote: > Dear All, > > Is there any new tutorial to install asterisk with SS7 support with dhadhi? > > > > >

Re: [asterisk-ss7] libss7 "dahdi show channels"

Hi use command core show channels verbose On Thu, Sep 24, 2009 at 9:38 PM, Farhan Q.Khan wrote: > You can check this command: > > asterisk -rx"core show channels" > > > == > > Regards, > > > Farhan > > -

Re: [asterisk-ss7] problem with libss7-1.0.2 installation; asterisk couldnt load libss7.so

Hi no need to install libpri for ss7 signaling only install libss7. On Fri, Sep 25, 2009 at 5:48 PM, Shuvro Hasan wrote: > Dear Salaque > > It is necessary to install libpri for ss7 signaling?? > > Suvro > > -- > *From:* Mohammad Salaque > *To:* aste

Re: [asterisk-ss7] Dahdi_ss7 e pri e1

no its not work only ss7 work On Thu, Dec 3, 2009 at 5:31 PM, Fabio Molinari wrote: > Hello everyone, > I configure chan_dahdi.conf with an SS7 link and a second link pri e1. Someone > has already done this type of configuration? > > > > > > > > ___ >

Re: [asterisk-ss7] Is Diguim Cards compatible with Solaris

yes digium card support all linux os u check ur karnnel . On Thu, Dec 10, 2009 at 3:10 PM, Amit Bhayani wrote: > Hi Guys, > > I have a requirement to have Zaptel / Dahdi Drivers installed on Solaris > and use Diguim Card on Solaris. Does Diguim officially support Solaris? > > Thanks a lot in adva

Re: [asterisk-ss7] Document on how to configure ss7 links in Asterisk

hi go step by step dahdi-linux dahdi-tool libss7 asterisk and fun ss7 is workinggg On Mon, Nov 30, 2009 at 5:12 PM, Gopalakrishnaiyer Venugopal-Q16770 < venui...@motorola.com> wrote: > Hi All, > > Kindly let me know how to configure ss7 E1 links in the Asterisk

Re: [asterisk-ss7] easy guide to install SS7 / Configure - 1 Link , 4E1s , CIC

hI FIRST INSTALL DAHDI-LINUX DAHDI-TOOL LIBSS7 ASTERISK and configure ss7 link dpc and opc its work On Sat, Oct 24, 2009 at 1:28 PM, TST Communication wrote: > I have a Digium TE420 card and Installed AsteriskNOW - 1.5.0 (64 Bit) (from > ISO Image), Q4 - Xeon server > > Can some one help

Re: [asterisk-ss7] asterisk server using two differents SS7 providers

Hi yes its possible use two opc and two dpc for each ss7 link. On Fri, Dec 18, 2009 at 2:02 AM, Daniel Pizarro Bustamante < dpiza...@yahoo.com> wrote: > Good afternoon, > > Is it possible to link one Asterisk box to two different providers (opc) > using chan_ss7? > > Regards, > > Daniel

Re: [asterisk-ss7] Link Not Aligned

hi your sustem.conf is wrong span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 try this On Wed, Dec 16, 2009 at 11:07 AM, Junaid Saeed Uppal wrote: > Hi All / Wolfgang, > > Both of the links came up [ configuration change to CRC4 on both sides and

Re: [asterisk-ss7] Link Not Aligned

Hi i have configure 4 ss7 link in single asterisk box. On Mon, Dec 21, 2009 at 1:46 AM, Vashkar wrote: > are you having two C7 configured ? I mean on ts 16 and ts 47 ? > > > On Fri, Dec 18, 2009 at 10:52 AM, bipin singh > wrote: > >> hi >>

Re: [asterisk-ss7] SS7 Configuration

Hi use libss7 its good . On Sat, Dec 19, 2009 at 12:33 PM, anwarul mamun wrote: > > Hi, > > I want to configure a voip gateway using asterisk and digium E1 card. > The inbound call is through internet and those call will be terminated using > E1 through ss7 link. > > Can someone please

Re: [asterisk-ss7] SS7 failover using chan_ss7

Hi yoherman use libss7 its work fine 2009/12/29 Yoherman > Hi all, > > i want to make SS7 failover with two asterisk boxes, each 4 port digium > card. > schannel at First and Fifth E1, time slot 16 > > > Telco - Asterisk A (ss7 port 1 (l1) channel 16) >| >+---

Re: [asterisk-ss7] how to put B-number ss7

Hi use the value in extensions.conf when u dial the number On Tue, Dec 29, 2009 at 3:30 AM, Alfredo Andaluz Prado < devilsoulbl...@gmail.com> wrote: > hi my ss7 privider need i put b number = C or # on the call any idea how to > put ? > > i have: > > DAHDI Version: 2.2.0.2 Echo Canceller

Re: [asterisk-ss7] Compiling chan_ss7 with Asterisk 1.6.0.21

Hi change the asterisk version use some new version of asterisk my ss7 working on asterisk-1.6.0 check it its work fine . On Sun, Jan 24, 2010 at 10:05 PM, Gregory Massel wrote: > Hello > > I can compile chan_ss7 1.3 cleanly against Asterisk 1.4.29, however, get > the > following errors when co

Re: [asterisk-ss7] How Asterisk + sigtran ?

Nacho, We can install n configure either Digium or Other commercial SS7 Solution for you. For more information please check my direct mail off the list. E-HORIZONS SOFTECH BIPIN RAGHUVANSHI +919310191818 OFFICE 011-32323262 On Thu, Jan 21, 2010 at 11:17 PM, Matthew Fredrickson wrote: > Nacho F

Re: [asterisk-ss7] chan_ss7 error - Write buffer full on CIC

hi use libss7 its work better than chan_ss7 On Wed, Jan 27, 2010 at 5:15 AM, Gregory Massel wrote: > > from chan_ss7 faq: > [snip] > > you can try: > > - newer version of chan_ss7 > > - jitter buffer option in ss7.conf > > [jitter] > > jbenable = yes > > jbmaxsize = 1000 > > jbresynct

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