Re: [asterisk-ss7] help on RLT Feature on libss7

We were able to implement Operator Redirect via RLT to our Nortel DMS-100. It was pretty much a hack but it worked for us. We completely changed how the FAR message is sent to the switch and can call it from the dialplan. -- _ -

[asterisk-ss7] help on RLT Feature on libss7

Can anyone suggest a way forward to implement RLT feature with asterisk and libss7? to accomplish this feature, we need to do the following: 1.Release the call with cause code(23). 2.Send the redirection number in the REL Message part on ISUP. I am able to release the call with cause code(23).

Re: [asterisk-ss7] Help

You can also use Sangoma A104d cards. I have purchased and used both digium and Sangoma cards. Thank you. On 11/9/11, bipin singh wrote: > Hi, > If you are new for asterisk then use only digium card . All > digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you > re

Re: [asterisk-ss7] Help

Hi, If you are new for asterisk then use only digium card . All digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you required good voice quality then use only echo cancellation digium card . We are using all card and work fine with ss7 link . On Tue, Nov 8, 2011 at

Re: [asterisk-ss7] Help

For SS7 you want a Digital T1/E1 card, which you can find here http://store.digium.com/products.php?category_id=2 Which card you pick is simply a matter of how many E1/T1 you want to support. Torrey On 8 November 2011 15:27, Rafael Machado wrote: > Greetings to all the list. > I make a purcha

Re: [asterisk-ss7] Help

I use TE410P for a year now, with 80-90 simultaneous calls, with no single problem. I even forgot what whas the card, lol. On Tue, Nov 8, 2011 at 6:27 PM, Rafael Machado wrote: > Greetings to all the list. > I make a purchase of a Digium card and the recommendation of friends would > like to indi

[asterisk-ss7] Help

Greetings to all the list. I make a purchase of a Digium card and the recommendation of friends would like to indicate what the best card to work with SS7 signaling. Can anyone direct me from experience? Rafael Machado Tel.: +55 (14) 3402 9702 Visite.: www.life.

Re: [asterisk-ss7] [Help] How to Show up the Asterisk Own Number/identity

Hi, First check the caller id you received NoOp(${CALLERID(all}). and second issue use Set(callerid=XXX) or talk to telco and set the default cli for ss7 link . On Mon, Feb 14, 2011 at 10:05 AM, Trung Nguyen Dac wrote: > Dear all. > > I have a trouble with making a

[asterisk-ss7] [Help] How to Show up the Asterisk Own Number/identity

Dear all. I have a trouble with making a call from Asterisk to a client (X-lite softphone in this case). When the client get an incoming call, it just show up the Asterisk's caller id ("Asterisk") . Could you pls tell me how to change the self identify of Asterisk. btw: when we connet Asterisk to

[asterisk-ss7] Help aligning 8 SS7 E1s with our Telco

Thanks Darren, Dave, Konstantin, all, We are now operational. Currently 2 spans and 5 more to come. *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mail

Re: [asterisk-ss7] Help aligning 8 SS7 E1s with our Telco

Hi José, have you checked that the Telco clocking is provided on the correct timeslot and that the SS7 signalling from the Telco is on the same timeslot as your Asterisk system is expecting? If you have an ISDN tester with audio monitor, you can listen for the presence of SS7 signalling (sounds l

Re: [asterisk-ss7] Help aligning 8 SS7 E1s with our Telco

2011/1/25 José Pablo Méndez Soto > Hi, > > We need assistance aligning 8 E1 ports (2x Digium Quad Span TE410P) with > our TELCO. > > We are working on a relatively big SS7 deployment for some time now. We are > using: > > asterisk-1.8.2.1 > asterisk-addons-1.6.2. > dahdi-linux-complete-2.4.0+2.4

[asterisk-ss7] Help aligning 8 SS7 E1s with our Telco

Hi, We need assistance aligning 8 E1 ports (2x Digium Quad Span TE410P) with our TELCO. We are working on a relatively big SS7 deployment for some time now. We are using: asterisk-1.8.2.1 asterisk-addons-1.6.2. dahdi-linux-complete-2.4.0+2.4.0 libss7-1.0.2 So far, we think that we should start

Re: [asterisk-ss7] Help with SS7 (No Audio)

Hi, Use asterisk-1.6.0 its working fine . On Mon, Nov 29, 2010 at 7:04 PM, Timothy Smith wrote: > Dear Users, > > I seeking help on with the asterisk+libss7. the call is successfully > setup but no audio either end. > > I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2, > chan_da

Re: [asterisk-ss7] Help with SS7 (No Audio)

problem and my CIC was not lined up.  My T1-1 was their >> > T1-5. >> > >> > >> > Thanks, >> > Dave >> > >> > -Original Message- >> > From: asterisk-ss7-boun...@lists.digium.com >> > [mailto:asterisk-ss7-boun...@li

Re: [asterisk-ss7] Help with SS7 (No Audio)

And, I'm not really familliar with libss7 config, BUT.. Don't you have to declare 4 links (or groups) under 1 linkset in your chan_dahdi.conf? in you current confi you have only 1 group. try something like this: linkset=1 group=1 sigchan=1 cicbeginswith=2 channels=2-31 group=2 sigchan= cicbeginsw

Re: [asterisk-ss7] Help with SS7 (No Audio)

> I had a similar problem and my CIC was not lined up.  My T1-1 was their > > T1-5. > > > > > > Thanks, > > Dave > > > > -Original Message- > > From: asterisk-ss7-boun...@lists.digium.com > > [mailto:asterisk-ss7-boun...@lists.digium.com

Re: [asterisk-ss7] Help with SS7 (No Audio)

vember 29, 2010 9:57 AM > To: asterisk-ss7@lists.digium.com > Subject: Re: [asterisk-ss7] Help with SS7 (No Audio) > > Thank you Gentlemen for your responses. > > I have done the dahdi_monitor, its only TX that has some input (see > sample output below). Thats for both outgoing and in

Re: [asterisk-ss7] Help with SS7 (No Audio)

Behalf Of Timothy Smith Sent: Monday, November 29, 2010 9:57 AM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] Help with SS7 (No Audio) Thank you Gentlemen for your responses. I have done the dahdi_monitor, its only TX that has some input (see sample output below). Thats for both

Re: [asterisk-ss7] Help with SS7 (No Audio)

Hi, I havent seen any useful information while loading and unloading chat_dahdi.so, however, I have attached below the output of ss7linktest and a debug during a call. By the way, what does it mean when RX is 0 (asterisk not receiving anything from the telco?) Thanks Tim ed channel

Re: [asterisk-ss7] Help with SS7 (No Audio)

do you have any relevant logs on asterisk console. set verbosity 3 unload chan_dahdi.so then load chan_dahdi.so you should see the ..cic expected on logs. try to set that cic as cicbeginswith. On Mon, Nov 29, 2010 at 7:56 PM, Timothy Smith wrote: > Thank you Gentlemen for your

Re: [asterisk-ss7] Help with SS7 (No Audio)

Thank you Gentlemen for your responses. I have done the dahdi_monitor, its only TX that has some input (see sample output below). Thats for both outgoing and incoming calls. How can I verify the circuit mapping? My core engineer (telco company) said that he is using the 1st channel for signalling

Re: [asterisk-ss7] Help with SS7 (No Audio)

I had similar problem BUT with chan_ss7. it was because my CIC mapping was wrong. On Mon, Nov 29, 2010 at 5:34 PM, Timothy Smith wrote: > Dear Users, > > I seeking help on with the asterisk+libss7. the call is successfully > setup but no audio either end. > > I am using Asterisk SVN-branch-1.6.

Re: [asterisk-ss7] Help with SS7 (No Audio)

Try sending a call via call file and see if you are getting both call legs. callchannel.sh #!/bin/bash echo "Channel: DAHDI/$1/$2 Callerid: $2 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: ss7 Application: Echo" > /var/spool/asterisk/tmp/test.call mv /var/spool/asterisk/tmp/test.call /var/sp

Re: [asterisk-ss7] Help with SS7 (No Audio)

Any particularly reason why your CIC begins with 2? I think they usually starts at 1. Check with the other side of the connection on CIC configurations. On Mon, Nov 29, 2010 at 8:57 AM, Abdul Basit wrote: > Try sending a call via call file and see if you are getting both call legs. > > callchan

[asterisk-ss7] Help with SS7 (No Audio)

Dear Users, I seeking help on with the asterisk+libss7.  the call is successfully setup but no audio either end. I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2, chan_dahdi.c is too bing but i can send it if required(perhaps to add p->dialing = 0. I didnt do it correctly?) I appreciate

[asterisk-ss7] help

From: "asterisk-ss7-requ...@lists.digium.com" To: asterisk-ss7@lists.digium.com Sent: Fri, March 26, 2010 8:27:57 AM Subject: asterisk-ss7 Digest, Vol 61, Issue 34 Send asterisk-ss7 mailing list submissions to asterisk-ss7@lists.digium.com To subscribe

Re: [asterisk-ss7] help for asterisk-ss7

zaptel.conf: loadzone = cn defaultzone=cn span=1,1,0,ccs,hdb3 bchan=1-31 # set this to 1-15,17-31 for E1 2009/10/17 zhubinyu > Hi all: > When i load chan_ss7.so,i get a error message like that,what is the > problem!? STAMP-ASTFIN*CLI> module load chan_ss7.so > > STAMP-ASTFIN*CLI> [Oct

Re: [asterisk-ss7] help for chan_ss7.so module load error!

; Is it miss some lib like libc.so.6,or something else!?thanks > > zhubinyu > 2009-10-16 > > 发件人： Nahuel Greco > 发送时间： 2009-10-15 21:23:18 > 收件人： asterisk-ss7 > 抄送： > 主题： Re: [asterisk-ss7] help for chan_ss7.so module load er

[asterisk-ss7] help for asterisk-ss7

Hi all: When i load chan_ss7.so,i get a error message like that,what is the problem!? STAMP-ASTFIN*CLI> module load chan_ss7.so STAMP-ASTFIN*CLI> [Oct 17 02:43:38] NOTICE[246]: config.c:625 load_config_link: Configured link 'l1' on linkset 'ls1', firstcic=1 [Oct 17 02:43:38] WARNING[246]: c

Re: [asterisk-ss7] help for chan_ss7.so module load error!

lib like libc.so.6,or something else!?thanks zhubinyu 2009-10-16 发件人： Nahuel Greco 发送时间： 2009-10-15 21:23:18 收件人： asterisk-ss7 抄送： 主题： Re: [asterisk-ss7] help for chan_ss7.so module load error! In the target platform check the output of: ldd /usr/lib/asterisk/modules/chan_ss7.so Saludos, Nahuel

Re: [asterisk-ss7] help for chan_ss7.so module load error!

In the target platform check the output of: ldd /usr/lib/asterisk/modules/chan_ss7.so Saludos, Nahuel Greco. On Thu, Oct 15, 2009 at 4:50 AM, zhubinyu wrote: > Hi all: >  I had compiled chan_ss7 support for asterisk run on blackfin uclinux > wiht blackfin uclinux toolchain.I get error l

[asterisk-ss7] help for chan_ss7.so module load error!

Hi all: I had compiled chan_ss7 support for asterisk run on blackfin uclinux wiht blackfin uclinux toolchain.I get error like that: STAMP-ASTFIN*CLI> module load chan_ss7.so [Oct 15 07:18:35] WARNING[215]: loader.c:364 load_dynamic_module: Error loading module 'chan_ss7.so': File not fo

Re: [asterisk-ss7] Help with the 'Location Number'

Thanks Matt Yes, I am familiar with C/C++ programming though not in a good shape at the moment. I will try my level best to get you online on MSN in your most convenient time (our time diff is +8hrs ) Regards Sam ___ --Bandwidth and Colocation Provid

Re: [asterisk-ss7] Help with the 'Location Number'

resea...@businesstz.com wrote: > Hi List > > I need to use ss7 optional parameter 'location number' on 911 (112) call > centre implementation to determine the location of the calling part as > passed by the mobile carrier Hey Sam. Sorry, I usually try to respond to most emails that nobody else h

[asterisk-ss7] Help with the 'Location Number'

Hi List I need to use ss7 optional parameter 'location number' on 911 (112) call centre implementation to determine the location of the calling part as passed by the mobile carrier Can someone assist to show how this information can be decoded and used in the dialplan. Thank you in advance Sam

Re: [asterisk-ss7] [HELP]:Suggestions needed in choosing proper channel Type in zaptel.conf for SS7 network

The easiest way to do this is to use libss7 + dahdi's native ss7 support, unless you want to look at all those repeated FISUs and LSSUs. Also this way you don't need to perform HDLC decoding, each MTP2 message comes as a single read/recv. Satish Chandra wrote: > Hi All, > > I am trying to develop

Re: [asterisk-ss7] [HELP]:Suggestions needed in choosing proper channel Type in zaptel.conf for SS7 network

"Satish Chandra" writes: > I am trying to develop a Tapping device that can be used in a SS7 network. I > am > not able decide on a particular channel type which I should configure in > zaptel.conf. The device needs to sit on a SS7 (T1/E1) link. As I will be > duplicating signal using a tapping

[asterisk-ss7] [HELP]:Suggestions needed in choosing proper channel Type in zaptel.conf for SS7 network

Hi All, I am trying to develop a Tapping device that can be used in a SS7 network. I am not able decide on a particular channel type which I should configure in zaptel.conf. The device needs to sit on a SS7 (T1/E1) link. As I will be duplicating signal using a tapping (Y-shaped) wire, I need not t

Re: [asterisk-ss7] Help with libss7 configuration

0P card in asterisk box. I thought you were using a TE410P as the converter. Regards, Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: July 16, 2008 3:32 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] Help w

Re: [asterisk-ss7] Help with libss7 configuration

TE410P as the converter. Regards, Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: July 16, 2008 3:32 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] Help with libss7 configuration Hi Christopher, Quoting

Re: [asterisk-ss7] Help with libss7 configuration

. I use the crossover cable to connect the two ss7 linksets together. Regards Mark. > > > Is this right ? > > > > Regards, > > > > Christopher > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mark Wilkinson > Sent: Ju

Re: [asterisk-ss7] Help with libss7 configuration

@lists.digium.com Subject: Re: [asterisk-ss7] Help with libss7 configuration Hi Christopher, I'm not sure I understand your question. What I have set up for testing SS7 is as follows :- SS7 Linkset 1 comes out of TE410P span 1, via a T1/E1 crossover cable into span 2 of the same TE410P t

Re: [asterisk-ss7] Help with libss7 configuration

Hi Christopher, I'm not sure I understand your question. What I have set up for testing SS7 is as follows :- SS7 Linkset 1 comes out of TE410P span 1, via a T1/E1 crossover cable into span 2 of the same TE410P to SS7 Linkset 2 Regards Mark. Christopher Srinivasa wrote: How were you able

Re: [asterisk-ss7] Help with libss7 configuration

How were you able to go from two SS7 linksets into the TE410P E1 card to a single ISDN after conversion on the crossover link? Did you not have traffic congestion problems when going from 2 to 1 ? Unless an SS7 linkset is equivalent to half an E1 ?

Re: [asterisk-ss7] Help with libss7 configuration

Anton: I was mistaken. At last test sig mtp2 did NOT work with Sangoma cards. The cards did work with sig = dchan. I'm going to delve further into this and identify why not. Apologies. -wasim On Thu, Jul 10, 2008 at 4:13 PM, Anton <[EMAIL PROTECTED]> wrote: > Wasim, > > As I understand, mtp2

Re: [asterisk-ss7] Help with libss7 configuration

Wasim, As I understand, mtp2 is hardwarely assisted sigchannel, which functionality is implemented in DIGIUM cards to act like dchan in PRI. I'd like just to confirm that we're speaking about the same, and if so SANGOMA must have such a support in their firmware too. As I know SANGOMA have ha

Re: [asterisk-ss7] Help with libss7 configuration

2008/7/10 Mark Wilkinson <[EMAIL PROTECTED]>: > Hi Anton, >I can only comment on it working with a Digium TE410P card > > Sorry, I don't have any Sangoma cards. > I can confirm it works with Sangoma cards as well as ofcourse, Digium cards. If you are using Sangoma cards, make sure you disabl

Re: [asterisk-ss7] Help with libss7 configuration

Hi Anton, I can only comment on it working with a Digium TE410P card Sorry, I don't have any Sangoma cards. Mark. Anton wrote: Is it only with DIGIUM cards or it works so with SANGOMA CARDS too? Anyone tried? On Thursday 10 July 2008 12:23, Mark Wilkinson wrote: Hi Matthew, I've just

Re: [asterisk-ss7] Help with libss7 configuration

Is it only with DIGIUM cards or it works so with SANGOMA CARDS too? Anyone tried? On Thursday 10 July 2008 12:23, Mark Wilkinson wrote: > Hi Matthew, > > I've just tested the latest svn version with 'mtp2' > instead of 'dchan' and my looped back linksets > come up without any problems. > > Regard

Re: [asterisk-ss7] Help with libss7 configuration

Hi Matthew, I've just tested the latest svn version with 'mtp2' instead of 'dchan' and my looped back linksets come up without any problems. Regards Mark Wilkinson. Matthew Fredrickson wrote: TCB wrote: I could not get C7 to get up with mtp2= on dahdi, works fine with the good ol' dchan

Re: [asterisk-ss7] Help with libss7 configuration

TCB wrote: > Thanks Matt, Any work around on the ss7_calling_nai issue. I have a > link that wont > terminate the calls but inbound works fine. > I need to understand exactly what the issue is. Can you explain to me what is wrong, and what needs to be fixed? Thanks :-) Matthew Fredrickson >

Re: [asterisk-ss7] Help with libss7 configuration

Thanks Matt, Any work around on the ss7_calling_nai issue. I have a link that wont terminate the calls but inbound works fine. On Wed, Jul 9, 2008 at 5:35 PM, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > TCB wrote: >> I could not get C7 to get up with mtp2= on dahdi, works fine with the >> g

Re: [asterisk-ss7] Help with libss7 configuration

TCB wrote: > I could not get C7 to get up with mtp2= on dahdi, works fine with the > good ol' dchan. Feel free to try. > > Just to let you guys know, I just fixed a bug that another developer introduced into the mtp2 code in DAHDI last night. If you update your DAHDI version to something new

Re: [asterisk-ss7] Help with libss7 configuration

Fantastic my linksets are now up ! Thanks for your help. Is this a known problem with dahdi ? Mark. TCB wrote: I could not get C7 to get up with mtp2= on dahdi, works fine with the good ol' dchan. Feel free to try. On Tue, Jul 8, 2008 at 9:03 AM, Mark Wilkinson <[EMAIL PROTECTED]> wrote:

Re: [asterisk-ss7] Help with libss7 configuration

I could not get C7 to get up with mtp2= on dahdi, works fine with the good ol' dchan. Feel free to try. On Tue, Jul 8, 2008 at 9:03 AM, Mark Wilkinson <[EMAIL PROTECTED]> wrote: > Hello, >I'm looking for some help configuring a system to test Asterisk's SS7 > setup. > > I want to have one syst

[asterisk-ss7] Help with libss7 configuration

Hello, I'm looking for some help configuring a system to test Asterisk's SS7 setup. I want to have one system setup with two linksets and a T1 cross-over cable between 2 ports on a TE410P (If this will work). At present, I have the following configured :- OS: CentOS 5 LibPRI, LibSS7, Dahd

Re: [asterisk-ss7] help-me, error in SS7

On Mon, 28 Aug 2006 17:33:47 -0300 Andre Luiz Martins Rodrigues <[EMAIL PROTECTED]> wrote: > Aug 28 17:00:38 WARNING[4432] config.c: Missing interface entries for > host 'gentoo1'. you have to add a line like if-1 => 123.123.123.123 to the host-section in ss7.conf, "if-1" is your network-device

[asterisk-ss7] help-me, error in SS7

Hello everyone, I have an Asterisk installed and I need to utilize SS7, however when I try raise the * appears the error: [chan_ss7.so] => (SS7 Protocol Support) == Parsing '/etc/asterisk/ss7.conf': Found -- SS7 channel unloaded. Ouch ... error while writing audio data: : Broken pipe Aug 28

Re: [asterisk-ss7] [help]no ringback tone and deteriorating audio...

Hi, I am now testing the chan_ss7 v 0.8.2 and the SUS and RES works great, Thanks to the Sifira guys :) But I still have a problem with the ring back tone when the caller is originating from the pstn side. I can see that asterisk is sending ALERT_CALL in progress but it seems that the Nortel

[asterisk-ss7] [help]no ringback tone and deteriorating audio...

march 8, 2006   hi all, i'm ryan and i've just joined this mailing list. i've tried installing chan_ss7 version 0.8.1 on my two servers. followed the step-by-step setup to test e1 cards with ss7 signalling that i found at voip-info.org. i was hoping somebody can help me with my problem here. i'v