similar to: One way audio problem on gateway to PSTN after some time, no NAT involved

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

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

Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an

> Is it possible to tell asterisk not to strip the leading 0 > of *incoming* MSNs? I use asterisk with i4l and whenever > I get a call from an long-distance party, the leading 0, which > should be there according the german numbering, is not. Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

Hi Well, subject says all. In Chapter 9.2.2.1, TBF, the parameter mpu or "minimum packet size" is explained as: > A zero-sized packet does not use zero bandwidth. For ethernet, no packet > uses less than 64 bytes. The Minimum Packet Unit determines the minimal > token usage for a packet. In my understanding an ethernet packet needs at least 14 (2*6+2) bytes or 54 bytes if

Hello everybody! After checking the complete wiki and the mailinglist archives I still haven't really found out why the following constellation does not work. We have an asterisk-System with some SIP-Phones and an old ISA Fritz-ISDN-Card used with i4l. The whole system is integrated in out (ISDN-)PBX for testing. The ISDN-Card is properly configured, as we are able to phone out, receive

Hi list! I've got a strange phenomen running asterisk for a while. After about two or three days without restarts, asterisk is not able to create SIP-Channels anymore, but gives me messages like Sep 4 00:12:06 WARNING[7175]: Unable to allocate channel structure Sep 4 00:12:06 NOTICE[7175]: Unable to create/find channel A reason this happens could be "hanging" SIP-Channels,

this : iptables -A FORWARD -i internal-interface -j ACCEPT iptables -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A FORWARD -j DROP doesn''t seem to work for active-ftp .. i even manualy loaded ip_conntrack_ftp but as u see it is unused : # lsmod Module Size Used by Not tainted ip_conntrack_ftp 4272 0 (unused) iptable_nat

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:

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 =

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300

hi, i'm added another patch to chan_ss7 it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/ New in version 0.10.1 (community version) - support for more than 256 channels - zap style addressing http://download.seiros.ru/SeirosPBX/chan_ss7/ http://www.freevoice.cz/chan_ss7/chan_ss7-0.10.1.tar.gz md5sum a3ca3031f8f4ef96d505be3b297b47cc

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, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Hello on Behalf Of idont know, Sangoma has a Media Gateway solution via SS7. They I believe are the only ones capable of connecting Asterisk via SS7. You may want to check them out. Heidi -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of idont know Sent: April 6, 2006 10:29 AM To: asterisk-biz@lists.digium.com

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

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

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