similar to: What Media Gateway (connected via SS7) do you use

Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server with Digium E1 cards ( 4 port cards). This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total

I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 =

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server. This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

When are 'we' going to have this solution Steve? :) You keep talking about it, and we keep asking when it's going to come about. I know myself, SS7 will be a make or break for our continued use of Asterisk. Even if we had some price indications would be good, and/or a timeframe? Don't want to seem pushy, but it's been on the cards for quite some time now. Ben -----Original

Hello everyone, I am trying to apply this<http://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diff>patch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors: Hunk #1 FAILED at 704. Hunk #2 FAILED at 715. I have done the patch modifications manually in l4isup.c There is just one question, how do I pass the RB file-to-play on an SS7 channel via

We'd be more than willing to test :) Our PRI provider has been trying to push us over to SS7 for ages. So we'd be very interested in assisting you. Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Underwood Sent: 12 January 2005 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion

I thought everyone should know this. Jorge, After reading your page in the http://voip-info.org/tiki-index.php?page=Asterisk+SS7 please advise Your U.S. customers that SS7 is not done the same way as in the rest of the world and the requirements are different. The U.S carrier's require 2 redundant links. I know this first hand because we run an SS7 network. CARDOSO Jorge Miguel wrote:

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in

I'm making outbound calls by placing call files in the asterisk outgoing directory. At times, the call would be hung by SS7 without even attempting (due to error in the outgoing number). I get the following on console: -- Attempting call on ss7/9297210213 for s at croom:1 (Retry 1) -- Sent IAM CIC=22 ANI=9134904821 DNI=9297210213 RNI= -- SS7 hangup 'SS7/callserver/22'

Hi, there are still some questions to be answered by OpenSS7.com in order to decide, whether E400P-SS7 is a good choice for the asterisk SS7 support. In the meanwhile I'm also in negotiations with another manufacturer (whose name I currently may not tell due to a NDA) of SS7 hardware, who gets likely persuaded to offer a cheap SS7-PCI-card which would be suitable for asterisk. Asterisk users

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

Hi all, I'm being requested to deploy an?IVR service?using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Sangoma a104 card have in product specyfication support for Line protocol SS7 , http://www.sangoma.com/products/p_aft-104-specs.htm [..] Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. [..] Anyone of you guys use line protocol SS7 for E1/T1 termination in asterisk ? As far I know asterisk don't have support for SS7 signaling, but my telco wants to setup

>I spoke with someone today who is interested in an IP Centrex solution that >starts with about 3500 extensions in a multi-tenant application. And >growing from there. > >I'm wondering about scalability of Asterisk. I'm trying to put my head >around how to put the whole thing together, if it can be put together. > >The nice thing about it is that if I can show

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page