similar to: Call Transfer over Fritz!-ISDN Card with i4l does not work

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

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

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

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,

> 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

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

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

Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN

Sorry for a slightly off-topic question, but I'm getting desperate. I tried to use asterisk to talk w/ ISDN line and tracked the problem to i4l. I have a Diva PCI 2.02 (S/T) card and am running RH8 w/ 2.4.20 kernel (first version that supports the 2.02 w/ hisax driver). The switch is a 5ESS running NI-1 protocol. The card sync's up and reports other traffic on the line, but fails to

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, Is it possible to use * as a gateway in the following setup: LAN (with Windows NT/Linux PCs) | Ethernet (IP) | Linux PC with * and AVM Fritz! ISDN Adapter | ISDN | Someone with a analog/digital phone (POTS) Basically, people sitting on their PCs will wear a headset, and whenever they want to call someone,

Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installing the drivers and which versions to use. Thanks, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -----Message d'origine----- De : vdasilva [mailto:aquamoon@wol.co.za] Envoy? : mardi 28 juin 2005 09:01

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

I can't find solution anywhere. I googled and find people with the same problem but there was no answers on how to fix this. I have W6692 based PCI cards that uses hisax driver (card type=36). Card is working fine under asterisk with i4l modem driver for incoming calls. If I want to dial out using some sip phone I don't get ring indication. Phone is ringing and I hear only silence until

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]

I have been trying to locate the patch that is supposed to cure the problem of hearing sound from the previous call when dialing through i4l and an hfc card. Does anyone have it? It is mentioned briefly in this post: http://lists.digium.com/pipermail/asterisk-users/2003-February/007530.html Thor