[asterisk-users] DTMF digits received, but not completely forwareded
Hello, we are running an Eicon Diva Server card with chan_capi and Asterisk-1.4.8. When we put in capi.conf softdtmf=off, the local command read() is recognizing dtmf digits from cell phone and from ISDN phones and from VoIP phones (via PSTN) very well, and asterisk is forwarding those digits correctly and completely to other switches via SIP. However, when an old analogue telephone is sending the DTMF digits, they are recognized approx 60% only. Even more strange: When we put softdtmf=on (relaxdtmf=off), the local command read is correctly recognizing _almost_any_ incoming DTMF digits from the Diva card. This is the desired behaviour, which we would like to keep. However, when the read() is running on another asterisk box, and the call is switched to that box using Dial(SIP/), only 80% of the digits are arriving at the other box. We tried with RFC2833 and with INFO (on both sides same). No difference. Can anyone please give me a hint, why not every digit, which the first box would recognize with read(), is forwarded by SIP to the other box? Why that difference? Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
Hello, escape the semicolons with a backslash! At least in astersik-1.6.X this works fine. I.e. replace in the SIP-Header-command all ; by \; Regards, Roger. Jonas Kellens schrieb: Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten = 10,1,NoOp() exten = 10,n,SIPAddHeader(Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) exten = 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12 14:56:19] -- Executing [...@from-test:1] NoOp(SIP/test6-0094, ) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-0094, Remote-Party-ID: eric sip:1...@192.168.1.150) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:3] Dial(SIP/test6-0094, SIP/test2) in new stack/ SIP debug : /asterisk*CLI sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] --- SIP read from 192.168.1.104:5063 --- INVITE sip:1...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: test 6 sip:te...@192.168.1.150;tag=adbbedf0959298ddo3 To: sip:1...@192.168.1.150 *Remote-Party-ID: test 6 sip:te...@192.168.1.150;screen=yes;party=calling* Call-ID: fb31bee7-94a6a...@192.168.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: test 6 sip:te...@192.168.1.104:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ In all the other SIP-messages there is no trace of the Remote-Party-ID header... Shouldn't there be a /*Remote-Party-ID: eric sip:1...@192.168.1.150;party=called */somewhere ?? Jonas. -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
Hello, the SIP header now should be sent. What the remote device is doing with this header, or whether the syntax of the header is as the remote device expects it, is another question. You can check with sip set debug on whether the header is now sent as you expect! If it does, I cannot tell you, why your Cisco device is not displaying it. Regards, Roger. Jonas Kellens schrieb: Roger, your answer did resolve something : /[Jul 12 15:51:24] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) in new stack/ However this SIP-header is never send as a SIP-message to the phone from where I'm placing the call. The name eric is not displayed on the screen. This is a Cisco SPA 941 and supports the Remote-Party-ID. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Hello, if the remote side (the public IP side) is capable to do something like asterisk's nat=yes (in sip.conf), than a mascerading router (like every cheap DSL router) would do enough NAT do let SIP work. If the remote side does not support that nat-hack (which is not SIP standard), than you will need a NATing router also doing a lot of SIP header rewriting. Maybe the most easy thing will be to install asterisk on the NATing machine and operating regular SIP links on both sides. Roger. Nivin Kumar schrieb: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Nivin Kumar schrieb: Is there a tool that will allow me to automatically change sip headers in realtime? Hi, imho changing the SIP headers will not be sufficient, since the old IP addresses are now private IP addresses (only in your network, outside, there are still public, but pointing not to your equipment). You will need a gateway, which does both: NAT 1:1, old IP addresses - new IP addresses and rewriting or all SIP headers, including those headers concerning the RTP endpoints. Maybe, you can do this with OpenSIPS. But I'm not sure about the SIP-headers for RTP. For H.323, it is imho less complicate, since it is robust for NAT and has no headers including IP addresses. Regards, Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early audio problem in chan_dahdi
... Hi, I've found the solution. I remembered, that with IAX2 - DAHDI everything is fine. Only SIP - DAHDI showed the problem. It seems, that chan_sip does not open ealry audio, if progressinband=yes in sip.conf. progressinband=no is needed for early audio. Strange! Anyway, that's ok for me now. Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Early audio problem in chan_dahdi
Hello, if have a problem since I switched to asterisk-1.6: When making an outgoing call through chan_dahdi, I cannot hear anymore early audio, the asterisk generated sound (as defined in indications.conf) is played. Thus, I cannot hear announcements by the operator, and when the line is busy, sometimes I can hear first the ringing indication by asterisk, and some moments later the busy. I already tried both in chan_dahdi.conf: callprogress = yes and callprogress = no No difference. What I'm doing wrong? Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI vs. PRI?
Ken D'Ambrosio schrieb: ... pretty pricey. Is there any reason that a BRI can't do exactly the same stuff, but on 2B+D instead of 23B+D? Hello, this depends on your operator and the telcom regulation in your country. In Germany, the main difference (besides the number of channels) is the numbering plan. With a BRI line, you'll get up to 10 single numbers, maybe consecutive, but without a mean to add or manage extensions. With at least two BRIs or with a PRI, in Germany you generally get a range of numbers, and you may manage the extensions on your own. E.g.: If you get the range -00 .. -29: Typically small business use than as numbering plan: -0 -10 .. -29 Maybe in your country the situation is similar. Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Virtual ISDN device /dev/XYZ
Hello, I do remember having read some weeks ago something about a virtual device provided by asterisk, behaving like an ISDN device, i.e. like /dev/isdn0. I know iaxmodem, but iaxmodem imho unfortunately does not transport raw ISDN data (HDLC frames), but only voice. Do I remember right, and there is an aseterisk application, providing such a device, which other linux executables can use, which expect a common ISDN device? Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangs up after 16 minutes on a call.
Hello, I had a similar problem with asterisk-1.2 long time ago, when I used the S(...) parameter in the dial command. Even if I used S(8), which is approx one day, asterisk hang up after exactly 64 seconds. When I erased the S() parameter completely, the problem was gone. Imho, this problem does not occour in asterisk 1.6. Anyway, if you are using the S paramter, try without and check, whether it helps! Roger. William Kenworthy schrieb: Hi, Ive just upgraded my home asterisk (on gentoo) from 1.4 to 1.6 and have an odd problem. After about 16 minutes on a call, it hangs up. Is -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma card reports HDLC errors
Hello, I've recently bought a new Sangoma A104d PCI-card. WANPIPE release is 3.4.7. Machine is a dual Xeon with Debian 5.0.3. Asterisk is 1.6.11 with recent libpri and dahdi. When I boot the machine (including hardware and wanpipe and dahdi drivers) and start asterisk, everthing runs fine for almost 5 minutes. Then every few seconds HDLC errors occour. I also tried booting - and starting asterisk later: The HDLC errors start almost 5 minutes after booting (not after starting asterisk). I cannot find any other job, starting after almost 5 minutes and consuming resources, top also show almost 100% idle time. I also tried to produce high load during the first 4 minutes, tar.gzing large directories. Anyway, no HDLC errors during the first 4 minutes, just after almost 5 minutes. For me, it seems, that there is anything within the sangoma drivers or the dahdi software overrunnig after appprox 4 minutes and 50 seconds, causing those HDLC errors. Any idea, who to find the cause of the error or how to solve it? Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma card reports HDLC errors
Hi, mea maxima culpa. I've found the cause, however Sangoma card and driver are working excellent. Just for the archive, to help someone else, maybe having the same problem: In order to test my system before bringing it to the data center, I plugged in a loop cable. Therefore I had to switch the E1 clock mode to master instead of normal. Of course I knew, I have to switch it back to normal, before connecting to the real carrier, but I forgot. Anyway, the dependence of the 4:50 minutes remains funny. Maybe a function of the divergence of the two clocks (my one the the carrier's one). Thank you answering anyway! Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_tool shows no alarms, but no line connected
Tzafrir Cohen schrieb: ... head -n1 /proc/dahdi/* # head -n1 /proc/dahdi/* == /proc/dahdi/1 == Span 1: WPE1/0 wanpipe1 card 0 (MASTER) HDB3/CCS/CRC4 == /proc/dahdi/2 == Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 == /proc/dahdi/3 == Span 3: WPE1/2 wanpipe3 card 2 HDB3/CCS/CRC4 == /proc/dahdi/4 == Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 If there are no alarms there, the wanpipe driver probably did not report them to DAHDI. Probably. But why? (When I turn of wanpipes, I can see them disapear with dahdi_tool.) How can I investigate the reason? Roger. -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi_tool shows no alarms, but no line connected
Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Connected | wanpipe2| AFT TE1 | N/A | Disconnected | wanpipe3| AFT TE1 | N/A | Disconnected | wanpipe4| AFT TE1 | N/A | Disconnected | However: # dahdi_scan 2 [2] active=yes alarms=OK description=wanpipe2 card 1 name=WPE1/1 manufacturer= devicetype= location= basechan=1 totchans=31 irq=0 type=digital- syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS Why are the dahdi tools not reflecting the values by wanrouter? Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6
Jeff LaCoursiere schrieb: Is it ready for prime time? He Jeff, at least version 1.6.0-beta9 was not yet very stable. We are also used to handle serveral Mmin/month with asterisk 1.4, but in our test environment, our asterisk 1.6.0-beta9 consumed file handles without releasing, and even a previous ulimit -n 9 could not prevent the system from causing network busies ... . Maybe, the current 1.6.0.3-rc1 has been improved. We also would like to merge the stability of current 1.4 with the new features of 1.6. Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
Hi, I figured out, that app_pppd suffered from overruns under high out traffic. (ping -s 600 destip was already high in this context.) I've just made a quick and dirty hack to fix it. If interested, just download the original package by Sirrix (as mentioned on VoIP-Wiki) and the replace their app_ppp.c by: http://planinternet.net/download/voip/asterisk/app_pppd.c Maybe I will later find the time to bundle a complete package, like the one by Sirrix. Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Packet size limit for HDLC?
Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with the analogue modems. Then, everything went fine (for a while). Unfortunately, PPP via ISDN is typically using a MTU and a MRU of 1500, and I found, that some commercial ISDN routers do not allow negotiating MTU and MRU. They insist to use a size of 1500. Since, using CAPI or ISDN4Linux (not via asterisk), pppd is working well with the MTU/MRU value of 1500, I assume, there is some packet size limitation in the asterisk part (including app_pppd). I tried to find any too small buffer or similar, but successless. May I ask you, where do you think, the limitation does come from: - from app_pppd (I don't think so) - from libpri - from chan_dahdi - from the dahdi kernel modules - from the asterisk kernel Any hint is welcome! Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
Eric \ManxPower\ Wieling schrieb: ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a Hi, the problem is, the other side (ISDN-router) does not negotiate the MTU while setting up PPP. I can see this in the log file: Our side is proposing 296, but the other answers with NACK and tells 296. I think, my side is doing something according RFC, when proposing a smaller MTU than usual, but this does not solve my problem, because: More info: http://www.znep.com/~marcs/mtu/ A MTU of 1500 is typical for PPP over HDLC, and when my solution does not do, what is typical, it is not compatible enough. Now I want bring asterisk and app_pppd also to work with a MTU of 1500 (like native linux ippp also does). I want to understand, why PPP via asterisk is failing, when MTU is 1500. Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] txfax_exec: Transmission loop error
Hi, I just installed Antonio Gallo's agx-ast-addons package in order to use app_txfax with asterisk-1.4. Compiling according to docs went well. However, I'm getting an error after the first page of fax: /usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec: Transmission loop error The (very first) page is transferred perfect anyway. Then app_txfax unfortunetly stops the transmission. Any hints? Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call interrupted after 64 seconds
Jaswinder Singh schrieb:
Can you post the part of your dialplan which causes this behaviour
Hi,
I've found, what's causing the problem:
My dialcommands are always of the type:
Dial(IAX2/user:[EMAIL
PROTECTED]/12345678,120,gS(${maxduration})M(connect^${some_params}))
or
Dial(SIP/[EMAIL PROTECTED],120,gS(${maxduration})M(connect^${some_params}))
${maxduration} is set to 86400 in most cases, sometimes to 3600
or 7200 (but never to 64). I checked this from within
the console.
When I leave the S() parameter away, there is no call, stopping after
64 secs. When I have the S() parameter, about every 10th call stops
after exactly 64 secs.
Thus, I assume a bug with the S() parameter in asterisk-1.4.x.
Can maybe someone check this on his machine, before I open a bug
report!
Roger.
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[asterisk-users] SIP call interrupted after 64 seconds
Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto get origin IP address from SIP call reliably
Hi, incoming SIP calls have a channel name in the form of: SIP/ip-adresss-of-peer-handle This is a way to get fetch the IP address of the remote side of a SIP call - in most cases. However, sometimes, instead of the IP address, there is a host name in the channel name. I assume, this value in the channel name is not the real IP address, but just a field filled in by the remote SIP client. Thus, this is not a reliable way to check the origin of a SIP call. What is a reliable way to read the real IP address of the origin of a SIP call? Regards, Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German SIP and/or IAX providers?
Peer Oliver Schmidt schrieb: ... as I am living in Germany, let me advise you against using VoIP providers in Germany. Most of the time they do work, but they are not as reliable as a regular phone company. Hi, on the one hand, I should ignore this thread, because it is not asterisk related, and risks to distribute commercial information, on the other hand, I think, at least one different opinion should be opposed to Peer Oliver's statement: I would agree, that there are big differences. Anyway, there are some VoIP providers who have specialiced in providing very reliable VoIP service to companies. ISDN technology is without any doubt (still) the most reliable thing in the telephony market. However, the difference to VoIP service offered by an appropriate company is imho almost negligible. Especially some small or medium sized providers offer very good customer support. Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip-info.org
Ed Nuñez schrieb: Is anyone else having trouble going into voip-info.org today? Yes. Me. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto use PRI lines (E1 or T1) for data calls?
Hi, I'm looking for a mean to send digital data over an E1 line, just like isdn4linux or Capi via AVM's FritzCard is able to do it with BRI lines (e.g. for PPP or ISDN raw connections). I'm not looking for modulated audio data representing digital data, like fax or the analogue modems of former times. I want an interface to the ISDN raw data, with an outgoing call marked as data, not voice. Best would by the behaviour of /dev/ttyIx. Thus I thought about iaxmodem. Could please anyone tell me, whether it is possible to make an ISDN data call using iaxmodem asterisk ISDN PRI line ? Thanks for any hint! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto use PRI lines (E1 or T1) for data calls?
Shane Spencer schrieb: point to point E1 lines? Or are you interfacing to a PSTN network for local calling/receiving? Hi, yes, PSTN. Normal operation is ordinary voice. Hm, the hybrid configuration mentioned in your link may serve as a workaround anyway. I should read this further. Thanks for the hint! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN config EWSD
Virmones Pereira Tavares de Miranda schrieb: How to configure asterisk and zaptel for ISDN ? EWSD? Hi, below is the ISDN part of my zaptel.conf. Imho crc4 is software selectable in EWSD, thus ask your provider! The D-channel could be found at another location, thus ask your provider! For T1 (or J1) links, the numbers of channels are different! (24 channels per span instead of 32.) Roger. ... # - PRI span 4 for E410P span=4,4,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Matthias Fechner schrieb: ... I use here mgetty+sendfax with a modem to receive and send fax messages. Is it possible to receive and send a fax with asterisk directly? Hi, did google for asterisk and fax show no results? Strange! Ok, what you need is Steve Underwood's package spandsp and the two asterisk applications app_rxfax and app_txfax, which is not included in spandsp nor in asterisk, but is generally to be downloaded near to spandsp. Regards, Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Matthias Fechner schrieb: ... yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Hi, ok, you want to use an external faxmodem? Something like that: outside (PSTN or anythin else) | V asterisk box | | (via analogue phone line) | V external faxmodem | | RS232 cable or similar | V PC with faxmodem support Yes, that's possible. You will need an ATA or in your asterisk box a card for the analogue phone line. Ok, what is your question now? Please ask more precisely! (Or is Yes, that's possible already the desired answer?) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk t38passthrough
Ricardo Carvalho schrieb: ... tries with the following codec preferences like G.711. On the other side there is PSTN, as I deliver my traffic in IP to a Telco that uses also Hi, that is not passthrough! You will need something to translate T.38 to one of the ordinary fax/modem-modulations, when switching to PSTN. Imho, this is not and will never be handled by asterisk's T.38 passthrough support. Anyway, Steve Underwood started to implement some T.38 support for his packages (spandsp/rxfax/txfax). Imho, this is, what you'll need. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zip code, city and area codes
Ronald Wiplinger schrieb: Is there a table available, which tells me if a zip code, city and area code matches? I doubt, that such a table does exist. Imho you will have to look for individual tables for each country. For Germany, look at: http://w3logistics.com Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Germany VOIP provider
Thameem Ansari schrieb: ... understand why do they give this 01801 if it is similar to local number? I am also exploring the options to get some landlines for flat rate. Hi, local numbers make the other party think to call someone in the respective area. Thus reaching someone at a local number, who has nothing to do with the respective local area, is spoofing. That's why 0180x numbers are an alternative. 0180x number do not implique any local relationship. But when you intend to move to Germany, you probably will have a local address. So what's the problem getting a local phone number? Some of the providers at http://voipliste.de/privat.html have also english web pages. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP reinvite _and_ NAT
Hi, I have a sipphone behind a router doing NAT, an asterisk box in the middle and another asterisk box, which works as gateway to further destinations. The asterisk box in the middle should do all call setup and tear down, but no RTP. RTP should flow directly between the sipphone via the router to the other asterisk box. When calling _from_ the sipphone, everthing is fine: The asterisk box in the middle is reinviting, and the other asterisk box is finally exchanging RTP with the sipphone in both directions. When calling _to_ the sipphone, there is a problem: The asterisk box in the middle again is reinviting, and the RTP stream from the sipphone than goes directly to the other asterisk box. But the RTP stream from the other asterisk box is sent to the private IP address of the sipphone (192.168) The NAT workaround is not effective in this case. Any hints? Just for understanding: Who is responsible in this case for the NAT workaround: The asterisk box in the middle who is reinviting or the originating asterisk box who is then sending the RTP traffic? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reinvite and NAT - Problem
Hi, I have following setup: ++ ++ | asterisk A |-| asterisk B |-- PSTN-gateways ... ++ ++ | .| | . =Router (NAT)=.== |. ++ . | SIP phone |- ++ asterisk A should do registration and call setup ..., and asterisk B should handle the media. Thus asterisk A should reinivite SIP phone and asterisk B on any call. I have asterisk 1.0.10, and on asterisk A the users are stored in mysql/sipfriends, which works fine. I already bugfixed in the source code, that chan_sip ignores the canreinvite- setting from sip.conf, and now calls from SIP phone to the PSTN gateway work perfekt: Reinvite occours, and using tcpdump I can see, that after call setup IP traffic is only between the router and asterisk B, but no more between router and asterisk A (besides hangup). The problem occours in the other direction, PSTN gateway to SIP phone: asterisk B is calling asterisk A, and than asterisk A is calling the SIP phone, as intended. Also as intended reinvite is taking place. But unfortunately, asterisk B is addressing the private (to be NATed) IP address of the SIP phone. Thus, audio data are flowing from SIP phone to asterisk B, but no audio data are flowing from asterisk B to SIP phone. The NAT workaround of asterisk is not working as desired. I assume, with a little source code modification the problem would be solved (like the sipfriends/canreinvite problem). Unfortunately I do not understand, who has to care about the NAT workaround. Is it asterisk A, who has to tell the right (SIP phones public) IP address to asterisk B (i.e. the one, where it gets IP traffic from instead of the one SIP phone tells), or is it asterisk B, who has to ignore the IP address, which SIP phone tells, but has to take the IP address, where traffic is coming from? Please explain how reinvite with NAT workaround should work! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1000s of extensions in one context?
Hi, is several 1000s of extensions in a context a problem? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] waitexten only provides one digit in chan_zap
Hi, I want to implement a lookup for valid extensions using agi. Thus I want chan_zap to accept some digits, then check via agi if the number is complete, run waitexten if necessary and check again ... Unfortunately waitexten only accepts one digit, regardless how may key strokes I did on my phone set. Even if I jump back in the dialplan, for asterisk passes again at the waitexten command, no more digits are accepted. Is waitexten no the right command to execute with chan_zap, an E1 line and overlap dialling? Is there something similar to misdn's waitfordigits, which I could use together with chan_zap? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1000s of extensions in one context?
Dovid Bender schrieb: several thousand extensions or several extensions called 1000 ? Several thousend extensions. exten = 497111234,1,goto(...) exten = 497111235X,1,goto(...) exten = 497111236XX,1,goto(...) exten = 497111237,1,goto(...) Several thousend extensions of maybe different length. For overlap dialing to operate correct (and no need to wait for timeouts) I would like to put the whole dial plan into the file extensions.conf. Before starting, I would like to know, whether there are experiences with such long dialplans. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No reinvite - reason?
Roger Schreiter schrieb: ... I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Hi, I've found the reason. Imho it has not yet been discussed here, and imho it is a bug in chan_sip. Thus, I will start a new thread, please see my next email! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug in chan_sip mysql support and canreinvite?
Hi, I did not yet study the newest chan_sip.c versions, but it seems, that chan_sip treats mysql-peers different from other peers, concerning the variable canreinvite. If this variable is not explicitely set for a peer or user in sip.conf, the global value for canreinvite in sip.conf is taken for this peer or user. This seems to be different for users defined via mysql table sipfriends. For those users canreivite in generally no. I just patched chan_sip.c (asterisk 1.0.10) to change this, and now reinvite works fine for me. Has anyone experiences with more recent asterisk versions (1.2.x) concerning reinvite for sipfriends users? Hasn't sipfriends been replaced by anything newer in 1.2.x? Thus, I assume, there is no need to open a bug notice in mantis. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP reinvite still does not occour
Hoa Thai Duy schrieb: Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Hi, yes, to avoid transcoding problems I only have one codec, just alaw. Anything else is disallowed. That's why I don't understand, why there is no reinvite. Thanks for answering! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP reinvite still does not occour
Hi, I have in my sip.conf disallow=all allow=alaw in order to avoid any codec problems disturbing reinvite. And of course I have: canreinvite=yes In extensions.conf there is only one Dial command. It has no qualifiers like t or T. Just Dial(SIP/[EMAIL PROTECTED]) Anyway, asterisk does not try to reinvite. asterisk tells -- Attempting native bridge of SIP/01234567 ... but in the debug output there no reinvite. Using tcpdump I can see, that the audio data are going via the asterisk box in the middle, not direct between the endpoints. Is there anything else, which can prevent a reinvite? dtmp-settings? nat-settings? Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No reinvite - reason?
BJ Weschke schrieb: ... I have no modifiers in my dial command. ... One reason might be is if you are passing parameters in app_dial (eg. Hi, sorry, I did use the wrong expression. No, there is no parameter like tT in the Dial command. I think, I've made everything according to the docs. Anyway: No reinvite and no idea how to find the reason. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No reinvite - reason?
Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find any attempt of a reinvite. Now I would like to know, why the asterisk box in the middle does not try to reinvite. Any suggestions? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP redirect
Hi, is it possible to let asterisk issue a SIP redirect? A SIP invite command by a SIP client should be answered by 30X Temporarly moved to SIP/ Is this possible with asterisk, maybe from within the dialplan? (reinvite is not what I'm looking for, because it does not completely release the originally called SIP server, e.g. if reinvite fails, ...) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with several SIP Providers (one way echo)
Alex Mosburger schrieb: ... SIPGate works also fine (better than SIPCall) but the ECHO is terrible. My side (* server) connected with X-Lite Softphone has a great quality, but the PSTN caller hears his voice with an echo. Did anybody already had such a problem? Do you think that my * server Hi, I assume some accoustic feedback (speaker - microphone) in your headset or what else you are using, or feedback inside your soundcard. (Some sound cards are able to feedback PCM-in to PCM-out. In this case, check, that rec-volume for that channel is 0!) In both cases, the gateway to PSTN (operated by your SIP provider or its partner) should reduce the echo notably, because they typically run bidirectional echo cancellers. Thus, look for a provider which combines good echo cancellation _and_ reliable signalling! Maybe the very cheapest provider is not the best one. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with several SIP Providers (one way echo)
... I forgot in my previous email one further issue: Maybe the ping round trip to your SIP provider and thus to the PSTN gateway is too long. The echo cancellation is typically limited to a reasonable echo run time. If the time is too long, echo cancellation will fail, because it would be too much, in most cases unnecessary, effort. Most DSL providers do offer fast path or similar things with improved ping times (e.g. below 30 ms instead of 70 ms). Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with several SIP Providers (one wayecho)
Alex Mosburger schrieb: ... It is not my end hearing or producing echo. My voice is heard correctly without any echo, but the other side hears his OWN voice several msec ... Yes, this is, what I meant. The other's voice is fed back by your device and running back to the other side. That's why the other side is hearing the echo. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID'S Romania - Bucharest
Oliver Vermeulen schrieb: ... We have ... Hi, I'm sure, there are a lot of providers of very interesting and useful and helpful products and offers reading and writing to this group - including our company. Nevertheless, noone is offering his products here, because it is not fair, if someone is offering and others are not, respecting this mailing list's rules. I don't know the right word in english, in german Oliver Vermeulen's behaviour is called unlauter, which means, that he is granting himself better chances by using forbidden means. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MWAnalyze question
James W. Brinkerhoff schrieb: ... My question for you is: When testing a tone as given in the example, Mwanalyze(8|8000|60|328), assuming there were no issues with the connection and the tone came through perfectly, what would you expect the 3 variables to hold? Hi, assuming slinear as the only transport codec and the other side is already sending 1 kHz when starting and also after ending, mwa_amplitude would be 2^15-1 (32767), mwa_ripple would be 0, and mwa_bad_timeslices would be 0. In real cases (good line quality, but codec G.711), mwa_amplitude is 29000 .. 31000 mwa_ripple is typically below 100, and mwa_bad_timeslices will count the cracks, and thus may be 0, if no jitter problems occour. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Responsecodes
Douglas Garstang schrieb:
Wow. If Asterisk could return SIP response codes that would be AWESOME.
... and the remote IP address (which may differ from the
address who registered).
Btw: Isn't the SIP response translated into a Q.931 code,
which can be read by ${HANGUPCAUSE}?
Roger.
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[Asterisk-Users] How to disable event_log?
Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to nice agi scripts?
Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts running with much lower priority than asterisk. Is there any mean to let AGI scripts run in a lower priority (except starting a new shell from the a short initial AGI script)? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over PRI
Michael Gaudette schrieb: I've been warned many times that Fax over VoIP is unreliable. How should I consider Fax over PRI channels with Asterisk? Is the quality and reliability good, or should I be prepared for alot of grief? Hi, we tested faxing a lot. With spandsp (app_rxfax/txfax) we had very reliable results, as long as only one or two simultanious faxes were transmitted or received. Than we bought a commercial DSP software, which is, according to the manufacturer, capable of sending or recieving up to 100 faxes simultaniously on a high perfomance machine. Well, we tested with a dual Pentium machine, not considerable as high perfomance machine, but we expected anyway good results. We could drive up to 30 fax sessions at once using a TE405 Digium card (Asterisk-1.0.10) without notable errors, more fax sessions at once showed a slightly increasing error rate. CPU usage was about 20% (80% idle time average). (We've sent in total 15000 faxes in order to get reliable mean values about the error rates.) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] slinear bandwidth
Anton Krall schrieb: ... Might be good for faxing though Hi, faxing suffers mainly from jitter, not from logarithmizing of the audio data. In PSTN G.711 is the standard and does not seem to impose problems on faxes. G.711 is ISDN quality! I assume, for fax quality and reliability the change between slin and G.711 will almost show no difference. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] slinear bandwidth
Anton Krall schrieb: Guys, how much bandwidth does slinear comsume and what quality can it be compared with? g711, gsm, g729? Hi, the bandwith is approx double compared to G.711, since it uses 16bit (signed) integers, whereas G.711 uses 8bit integers. The human ear has approximately a logarithmic sensivity, and G.711 has a logarithmic resolution. Thus the quality gain of slinear compared to G.711 may not be very much. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Milliwatt Analyzer available
Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple of 0.5 ms, thus the valid frequences are: 2000 Hz, 1000 Hz, 666.7 Hz, 500 Hz, ... Furthermore the application computes the ripple on that tone. In order to detect audiogaps and short noise on the line, one can define a treshold and a timeslice duration (typically 1s to 0.1s), and the application will compute the ripple for each timeslice and count the timeslices with a ripple greater than the given treshold. Thus the application is a tool to verify the line quality, e.g. for least-cost-but-not-too-bad-line routings. For conveniance Mwanalyze also generates a tone of the frequency it analyzes. Thus a bidirectional operation, and test for frequencies other than Milliwatt's 1000 Hz are possible. Anyway Milliwatt is much much more economic to CPU and RAM! For details see inline documenation or output while loading the module app_mwanalyze.so! Now, I will try to contact to dev-list, in order to put this application to future releases. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote IP address in channel?
Hi, when I get a SIP call from an unknown user, I can see the IP address in the channel name. When the call comes from a known user (sip friend), I can see only the username in the channel name. Ok, most users will use the IP address, which they also register, thus can be lookup up in the registry. But I'm looking for a reliable channel variable or any reliable mean, where a SIP call from a known user really comes from. Any ideas? Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test call quality
amaury BOSSE schrieb: Is there a free linux tool which can test voip call quality between two Asterisk PBX. It will help me to test the WAN network between them. I have only found commercials ones, so if you know a free one, let me know. Hi, just some hours ago I published in this list: http://www.planinternet.net/download/voip/asterisk/app_mwanalyze.c Copy it in the asterisk source into apps/, and add it in apps/Makefile, and run make, ! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Milliwatt Analyzer available
Juan Carlos Castro y Castro schrieb: Could I use this to distinguish human voices from machine beeps and/or ambient noise etc, by (after a few adaptations) calling it a number of times on the same set of samples with some representative set of frequencies? Or is there a better, less CPU-torturing way to do that? Hi, I doubt, Mwanalyze will be versatile enough for that. I think, that would be another project. The only case, you could profit from Mwanalyze for your purpose, would be, if the machine noice/beep/tone will have _either_ a lot _or_ none portion of exectly one of the frequencies supported by Mwanalyze. Mwanalyze will e.g. tell you, that you have an amplitude of 300 (slin) of 500 Hz and a rest of another 1 (slin). This would be an indicator for a synthetic 500 Hz tone. Whereas a 500 Hz amplitude of 0 would be an indicator for the absence of natural voice sources. Maybe under certain conditions this will be sufficient for your purpose. In general you will need a spectral analysis of (almost) the whole auditible spectral range. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Matching '*'
Douglas Garstang schrieb: I'm trying to find a way in extensions.conf to match ANYTHING dialled, Hi, your subject is probably not correct. You want to catch anything except h, t, ...? Maybe you want to get matched the digits and *. Thus try: _[*0-9]. This will match any dialed string, which starts with * or a digit and has at least a length of 2. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Channel Status
Douglas Garstang schrieb: If dial() doesn't return until after the call completes, it means the channel status AGI command is a waste of time. Hi, you are right, dial will block, so you won't get the channel status by that method when having an outbound call. You can use the manager. But will have to poll. To avoid polling, I tried to use the manager and parsing the events, but unfortunately the events seems not be reported very reliably in the manager. On high load, some link events imho get lost and are not reported. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Scripts Terminate too Soon
Douglas Garstang schrieb: ... HOWEVER, if the CALLER hangs up the call, it seems Hi, did you try the dial command option g? I did not neither, but when I understand the voip-wiki right, it might help you. Roger. Voip-wiki page about dial: http://www.voip-info.org/wiki-Asterisk+cmd+Dial ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
Matt Roth wrote: ... What is being discussed here is basically what I was planning on ... This sounds like a programming project. Something like a stripped down softphone (or possibly a plugin to an existing phone) with Hi, since I need rather a tool not that versatile but within some days than a very nice tool available in some weeks, I've started to code a little tool, which is just doing some analysis for one given (but variable) frequency. Thus it can be used as opposite for the Milliwatt application. For some convenience it also produces a tone, of that frequency, it is analyzing. I've called the new application Mwanalyze. E.g. Mwanalyze(16,8000,60,328) is producing a 500 Hz tone. For a first look, fetch: http://www.planinternet.net/download/voip/asterisk/app_mwanalyze.c Docu inline. It does not yet perform any analysis, but already generates the tone. I will tell here, when finished. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to reset Digium card while asterisk is running?
Hi, I currently have a yellow/red alarm on one span of a Digium card. It is not the first time, this already happened some months ago, and I expect to clear the alarm when rmmoding and insmoding the zaptel and wct4xxp modules. Unfortunately I can't rmmod while asterisk is running and I can't stop asterisk, because of lot of traffic on other channels. Ok, I could wait until 2 o'clock in the night, in order to shut down asterisk on low traffic. I wonder, if there is another way to reset a Digium card without the need to shutdown asterisk? Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analyzer for Milliwatt
Hi, app_milliwatt is a nice tool for a quick check of the line quality. Anyway, hearing to that tone for more than a minute is painful. Does anyone know the opposite application, i.e. an application, that hears and listens for a 1000 Hz tone and displays the quality in any unit? If not, I'll think about developing one. Regards, Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to messure PDDs, how to detect fast hangup?
Hi,
how can I know in the dialplan, whether and when I received
the ringing event?
Imho, the only way is to parse all events using the manager
and to forward this information by an application to the
dialplan. The application would have to be called on
connect or on hangup.
Hints for a more simple way are welcome!
Does anybody think about a new state in ${DIALSTATUS}:
FAST_HANGUP (for a hangup before ringing)?
Would anybody else appreciate a dialplan
variable ${RINGTIME} like DIALEDTIME and ANSWEREDTIME,
but indicating, when we received the ringing-notification?
Roger.
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[Asterisk-Users] How uniqueids are formed - possible race conditions for linked channels?
Hi, the uniqueid obviously consists of a timestamp part and an continously incremented part, separated by a dot. The two channels of a call in most cases have the same number before the dot (timestamp) and consecutive numbers after the dot. Now I wonder, whether I can rely on that scheme. I assume, the timestamp part can be different, e.g. if between the creation of the incoming channel and the creation of the outgoing channel the system clock switches to the next second. (Or even more, if an AGI- script or anything else has consumed more time in between.) What about the part after the dot? Is it possible, that the outgoing channels gets there a higher number than the one from the incoming incremented by 1? May e.g. another incoming channel at almost exact the same time get that number instead? Thanks for some hints about that! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How uniqueids are formed - possible race conditions for linked channels?
Stefan Reuter schrieb: To propose the best solution we must know more about your actual use case. Thanks for your answer! I want to track the ringing event of the outgoing channel. Unfortunatelly the link event is fired not before connect. Thus, I see first the incoming channel (SIP or IAX) which tells me the username, who is using the channel, and some more data, which identifies the call uniqueliy for my system. Afterwords, I see the outgoing channel (Zap, SIP or IAX), which after some time (up to 20 seconds for mobile channels or even longer for bad routes) gets the ringing event. Now I'm looking for mean, to get the ringing event (of the outgoing channel) related to the incoming channel name. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How uniqueids are formed - possible race conditions for linked channels?
Stefan Reuter schrieb: ... So having a look at Asterisk 1.2-beta2 is probably the way to go. Great! Yes, this will solve my problem. Let's upgrade ... Thanks! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 with Asterisk
Usman schrieb: anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... google shows for the words asterisk and ss7 amongst the top five: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 This will give you the desired answers. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Faxdetection in IAX? (Missing audio samples)
Hi, please don't bother me continuing trying to fax, even if I've got convinced, that it generally won't work! I've found a strange behaviour, when sending from IAXCLIENT1 - asterisk - IAXCLIENT2 or from IAXCLIENT1 - asterisk - SIPCLIENT2 When IAXCLIENT1 is sending an absolutely constant frequence (fax detection tones at the very beginning of every fax session), audio samples get cleared (set to 0). I dumped both, the outgoing samples from IAXCLIENT1 and incoming from IAXCLIENT2, respectively heard by myself with a SIP phone. IAXCLIENT1 sending a constant tone of exactly 3000 msec, which leads in the dump file to a very periodic pattern of hex numbers, and thus is easy to find manually in the dump file. Arriving at IAXCLIENT2 respectively at the SIP phone is that tone for exactly 500 msec, afterwords silence for approx 2450 msec. Is there an explanation for this behaviour? Thanks for any hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing - at astricon ?
Kresimir Petrovic schrieb: ... What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... Hi, it is rather a question of IP quality than good luck. I think, 99.9% of all faxes are transported via G.711. Is there any telecom network operator left using ananlog lines? Analog is either the 5 meter way from the fax machine to the BRI/PRI adapter at the wall or maybe even the 1000 meter way from the home/office to the next switch of the network operator. Thus the main difference is the high quality of the managed ISDN network compared to the unmanaged IP network. That's why, imho, analog faxing via VoIP is not pure luck, since one _does_ know the relevant parameter. If I have some means for a very good internet connectivity, faxing will work without problems, if not, you are right, it will become pure luck or just impossible (without T.38). Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing - at astricon ?
Roy Sigurd Karlsbakk schrieb: ... see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... Hi, maybe one should update this link. I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) Since there are more and more regions in the world, where internet connectivity quality approaches to ISDN quality, analog faxing over VoIP becomes reliable and hassle free. You should have 128kbit in both directions, better 256kbit, maybe some QoS build in your router (e.g. Linux's iproute2), and pingtimes below 20ms to the VoIP-provider (PSTN-gateway). DSL with fastpath or internet by TV cable does provide this standard imho and become more and more available. Thus we shouldn't discourage people generally of faxing, even if there are a lot of trouble reports. Who can count the success stories with (analog) fax over IP, which are not posted? As far as I see, there are more users faxing without observing quality differences to ISDN than users with problems with fax over VoIP. This is, what various partners of ours do report after having replaced BRI connections by VoIP in some small and middle sized companies. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing
Carlos Alperin schrieb: ... Bank, and send all the faxes through that system, and forget to try anything through Asterisk. We're already tired of the complainings from customers that never can send faxes, or sometimes some pages. Wow, strong words! Since our experiences with asterisk are mainly positive, we won't forget faxing through asterisk. As long as asterisk is just used as pbx or switch, it is obviously capable of transporting 100 and more faxes simultaniously without problems. The only thing missing seems to be a reliable modulator/ demodulator and T.30 machine. rxfax/txfax+spandsp is still far away from being reliable under high load, but developing. Our experience is, that it does a very good job when sending or receiving single faxes. As soon as three or more faxes are processed at once, problems do occour. We are currently testing a commercial fax library with the goal, to implement an IAX client as fax server, in order the send or receive some dozen faxes simultaniously. It is too early to say, if we'll succeed, but we hope so, and thus won't forget faxing through asterisk. This is _no_ announcement of a product, since we currently won't have the right to distribute the library, but we think, it would be very interesting to know, whether it works or not. Those, looking for a GNU licenced tool will have to wait for progresses of spandsp anyway. By the way, the producer of the library announces T.38 support in some weeks. asterisk just started to support T.38 pass through, and probably will do it reliably in some weeks. Maybe this will enable T.38 support for asterisk - though unfortunately not for free nor open source. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing
Andy Kuo schrieb: ... It sounds like Asterisk should handle T.38 fax (at least at very light ... Can you please tell me what ATA you are using? Could it be that the Hi, are you sure, your ATA does support T.38? If so, you'll need the most recent CVS versions of asterisk, and rather consider yourself as tester than expecting a reliable support of T.38 passing through. Imho T.38 passing through won't be interfered by other traffic on the same gateway. It will work or not, but not very depending on system load. Maybe you are talking about common (ananlog) faxing. I tested the IAXy, a Sipura ATA and the FritzboxFon. The latter is one of the best ATAs I ever saw. All those three enable faxing under the condition, the ping round trip to the gateway to PSTN is not too long. Below 20ms should be reasonable. The quality and reliability of this setup is excellent. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing
Lee Howard schrieb: ... See: https://sourceforge.net/projects/iaxmodem/ Hi, great! I googled a lot, before we started our project. A iaxclient which works as modem (aka tty device) would be great. Do you think, one can connect a hylafax server to iaxmodem? Yes, I know, that jittering will be a problem and will make necessary some buffering, but not too long, ... Well, we will proceed the project anyway, to see, whether it is possible or not. We have a lot of clients, reporting, that faxing through our asterisk gateway works very well. Our clients are VoIP providers, thus the typical setup is: fax -analog- ATAbox -SIP- asterisk1 -IAX- asterisk2 -PSTN- remotefax or asterisk1_with_spandsp -IAX- asterisk2 -PSTN- remotefax asterisk1 is the machine of the providers, which use our termination service. asterisk2 is our gateway. We have one customer, whose customers are SOHOs with typical small PBXs, supporting 2 or 4 lines. He now puts the FritzboxFon between PSTN and their PBXs, in order to take the outbound phone traffic via VoIP, including fax. He is using as internet connectivity KabelBW, a regional cable TV provider. His setups are success stories. Thus, I assume, IAX and the jitter won't be the problem, when the internet connectivity is good enough. Well, we'll proceed our tests and see. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing
Lee Howard schrieb: ... to go through all of the effort of doing it yourself. But you actually seem to be looking for something beyond just knowing if it's possible or Hi, yes, we consider offering our PSTN-gateways to fax providers during off peak times, in order to get a higher average load on our machines. Interconnections to telcos are costy, and thus some more off peak traffic would help. So we have to check, whether it is possible, which solutions are suitable, and what each solution will cost. Unfortunately our budget is currently very limited. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connect not signalled (SIP - Zap)
Hi, I've had a strange problem several times during the last days: A call is established, both parties have audio in both directions, but asterisk is still waiting for connect. Thus after timeout (120secs) the call is terminated with either busy or no answer. This is annoying for the both parties, who are already speaking, because they get interrupted. In the cdr I can find the call with 120 secs duration, 0 secs billsec. Has anyone had a similar problem so far? Or any ideas? Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS using a PRI channel
Hi, I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. Now, I want to do the same using asterisk and a digium PRI card. I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote end will become signalled a voice call, and imho the SMSC wouldn't answer a voice call, but expects data calls. Well, originating succeeded, and the respective context in the dialplan was accessed: -- Executing SMS(Zap/94-1, me||mycellnr|Test) in new stack -- Executing NoOp(Zap/94-1, Done) in new stack The application SMS returned without error, but returned immedeately (much less than 1 sec.). Of course, no SMS was sent. How can I debug this? How can I force Zap to data mode. The d option seem to be something different. Did anybody try sending SMS to german Vodafone or other SMSC mentioned in the smsclient package? Thanks for hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Txfax
Chris Shipman schrieb: What build of SpanDSP did you use? spandsp-0.0.2pre18 I'm working on a windows program so users can print to a local printer which will be forwarded to the asterisk server to be faxed. So far the program FTPs a Tiff to the Asterisk server to be faxed with a Sample.Call file.(For lack of a better method thus far) I don't understand, what you are telling or asking us with this information. Has it something to do with your question? If not, please avoid confusing with additional infos which are not relevant! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pass through of T.38
Hi, I found some contradicting infos about pass through of T.38 data. Are there any experiences of just passing T.38 via SIP from one T.38 application or gateway trough asterisk to another T.38 application or gateway? Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip (without understanding the content)? Please tell me, if you have knowledges or experiences on this topic! Othervice, and if I won't find further reliable information saying it cannot work, I'll try it. And of course I will report the results later here. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Txfax
Il Neofita schrieb:
Is there some way to know if the fax was received correctly or not?
Hi,
after also having asked here some days ago without an answer,
I assume, there is currently no way.
Thus, I started to study a little bit the source code.
Maybe I found a solution, which will work.
I haven't yet understood the whole souce, so I don't know,
whether my solution was silly or not or whether my solution
is reliable or not.
For me it is working great, but use with care!
Roger.
My proposal for a fax-send-success-feedback:
1. Look in txfax.c (approx line 60) for the function
static void phase_e_handler(t30_state_t *s
2. Add some code, which writes the number of successfully sent pages to
an channel variable:
static void phase_e_handler(t30_state_t *s, void *user_data, int result)
{
struct ast_channel *chan;
char far_ident[21];
char minibuf[11];new line
chan = (struct ast_channel *) user_data;
if (result)
{
fax_get_far_ident(s, far_ident);
pbx_builtin_setvar_helper(chan, REMOTESTATIONID, far_ident);
sprintf(minibuf, %d, s-t4.pages_transferred); new
pbx_builtin_setvar_helper(chan, FAX_PAGES_TRANSFERRED, minibuf);
new
Then you'll find the variable ${FAX_PAGES_TRANSFERRED} accessible from
the dialplan.
If you know the number of pages to be sent, you can compare and see if
complete.
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[Asterisk-Users] Max concurrent faxes with txfax/spandsp?
Hi, I tried to use txfax to send several faxes at the same time. It seams, that one can't send more than 3 faxes at once, or one risks to get 50% and more aborted faxes due to errors. The CPU usage is below 97%. I tried with Opteron and IntelP4: same result. Ok, I know, that faxing via a digital line is complicate, and I shouldn't complain, but I would like to know, whether these are typical values or whether one could increase the max fax number by any means? Maybe force to a slower, but more error proof modulation? Regards, Roger. P.S. After faxing approx 100 faxes, CLI show channels shows a lot of channels, which seam to be forgotten, not hangup faxlines. Is this a known weakness of txfax? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to tell reason for hangup or busy in SIP or IAX
Hi, using Zap, I have several messages to pass when terminating a successful or unsuccessfull call, indicating the reason e.g., why a call failed. Using SIP or IAX2, I know only Hangup Busy Congestion without passing any more detailed information. Am I right, that I can't tell the caller in SIP or IAX2, whether the call was rejected or whether the called number does not exist. I have to assign those hangupcauses to one of Hangup (which I use for normal termination after successfull call and for No answer after timeout. Busy (which I use, if the called phone is busy) Congestion (which I use, if our gateway or one of the used carriers are busy) Thanks for any hints, if there further means! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debug info from txfax - howto?
Hi, I read here in this mailing list about the debug info from txfax. I plaid a lot, but didn't get debug infos. I added the debug argument to txfax, I enabled debug in logger.conf, what else should I do? In which file or medium can I then expect the debug infos from txfax? Thanks for hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Return code of txfax
Hi, I have asterisk 1.0.7 and spandsp-0.0.2_pre18. txfax return a non-zero return code only if the fax file is not found. Unfortunately I can't get any information, whether the fax was transmitted completely or not. Will an update to a newer version change this? Thanks for telling me your experience! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?
Asterisk schrieb: I'm looking to develop some custom AGI that will be MySQL intensive. It appears Asterisk supports many different development environments. Which would be best suited for Asterisk and MySQL? Hi, this is the same question as for other applications: Use C, if you want to optimize runtime, use Perl or PHP if you need fast developement. I've developed AGIs in C as well as in Perl. Both languages are suited well, with its respective advantages and disadvantages. Others report good experience with PHP, Python, and some are even using Java. The interface to asterisk is just stdin and stdout and a little parsing. So don't worry too much whether your preferred programming language is suitable for AGI programming! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P + SPANDSP fax problem
Ma Zhiyong schrieb: ... Trace shows that the fax is received successfully. Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX(Zap/94-1, Hi, sorry, I don't know the solution to your problem, but I would like to know, how did you get that trace? I'm looking for a reliable way to determine, whether TxFax did send a fax completely. I also tried the option debug, but never saw such a trace. Which version of spandsp are you using? Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TxFax - RxFax on same machine hangs
Hi, I noticed a strange behaviour: Faxing using spandsp (TxFax) from my asterisk box to my old, common fax machine at home works fine. Faxing from the same box to my office pc-fax (Hylafax) also worke fine. Receveiving faxes on my asterisk box using spandsp (RxFax) also works fine. It is a PSTN number connected to the digium card of that asterisk box. Then I faxed from my asterisk box (TxFax) to that PSTN number, my asterisk box answers the call, TxFax and RxFax, both start, but no tif file is created in the directory, where normally the fax files are created. Using show channel I can still see both apps, TxFax and RxFax, even after half an hour. Then I stopped using soft hangup, and tried again several times: Same result. So, do I have to avoid to fax to myself in any case, because I risk to produce those never terminating jobs, which probably consume some resources? How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Result from TxFax
Hi, there are some messages indicating, that TxFax is able to return -1 on failure. Well, I tried a lot but didn't succeed. I even sent a fax to a phone set, picked up the hand set and waited until timeout of TxFax. There is no difference to success. The only thing I could determine, is, when the other party hangs up. Any other case called the next priority in the dialplan. Is there any reliable mean, to check, whether a fax is really sent successfully and complete? Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFax - RxFax on same machine hangs
Steve Underwood schrieb: ... If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will Hi, the setup is: TxFax (Box A) Dial(Zap...) (Box A, Digium Card) v PSTN v Box B, Digium Card Dial(IAX2...) (Box B) v RxFax (Box A) TxFax and RxFax ran on Box A. The PSTN call was accepted at Box B and then forwarded via IAX2 to Box A. RxFax and TxFax did nothing, and were never terminated, and thus needed an expicit Hangup command. Regards, Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 decoding
Hi, I searched a while about T.38 decoding, and learned about the bounty for T.38 support for asterisk and some softdecoders and some hardware de- and encoding T.38. Now I wonder, if there is already any (almost) ready to use solution for decoding of T.38 faxes? My szenario would be: - Receiving a SIP call (containing the T.38 fax) by my provider with my asterisk box. - asterisk would forward that SIP call to the converter. - The converter would send the SIP call back to my asterisk box, but now with the fax deocoded to an ordenary anolog fax. Has anyone experience with a working solution, maybe a foreign service provider doing it, or a working (asterisk independent) software? Thanks for any hints! Roger. P.S. Currently I'm trying to understand, what ionidea's T.38 software is already able to do, but I'm still confused. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opteron Hardware with Asterisk
Hi, asterisk compiled fine and is running very stable on our dual opteron in 64 bit mode. When loading G.729 library we have to peload libz manually for any reason, but besides that minor issue, everthing is fine. We didn't yet test the limits of that machine. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM gateway hardware
Allan Kamau schrieb: ... I am looking for a GSM VoIP gateway for use with Hi, do you think of something to interconnect to GSM carriers via cable (GSM-A) or do you think about using a GSM-modem with all its limitations? For the first option I could forward your email address to someone providing GSM-A stacks for asterisk. For the second option, it might be interesting for you, that we are currently also working on asterisk support for a GSM-modem. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manger-command Getvar?
Hi, I'm trying to use the manager cmd Getvar. Unfortunately I always get (null) as variable content. I'm using asterisk 1.0.7 When calling a non existant channel, I get an appropriate result. This is what I tried and got: Action: Getvar Channel: SIP/01234567-5242 Variable: CALLERID Response: Success CALLERID: (null) Any hints? Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy tone (German/Dutch/French)
Hi, if I understand right, the best way to indicate a PSTN line busy, is something like that in extension.conf: ... background(busy-tone) ... busy So the caller will first hear my busy-tone, and after some seconds, when PSTN honours the busy indication (cmd busy), he hears the busy sound by his network operator. Are there any files like busy-tone.gsm with german (aka french or dutch) busy sounds? Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp
Marco Parmeggiani wrote: ... i had no problems receiving faxes with version 3.7.2. on the other hand i have big problems in sending multipage faxes. only Hi, where did you get that version? On libtiff.org, 3.6.1 is the most recent one. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: rxfax problem - libspandsp issue?
They keep breaking the FAX support in libtiff. 3.6.1 is broken, a ... Hi, thanks for the information about libtiff 3.6.1. I had to search a while in order to find the old libtiff 3.5.7, which now works fine in my asterisk installation. For those being in the need of libtiff 3.5.7: Pay attention to my subsequent email! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp
Hi, package tiff-v3.5.7 contains the currently recommended version of libtiff in order to run spandsp (fax support for asterisk). Imho tiff-v3.5.7 is not very easy to find in the internet, and maybe will almost disappear, because it is an old version, I put it on our little asterisk download page. Maybe it'll help someone. It works fine together with the other asterisk stuff (around version 1.0.7) located in that directory: http://planinternet.net/download/voip/asterisk Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Miax: Digital voice channel when connecting to asterisk
Hi, I've bought a Siemens GSM-modem based on the Siemens TC35-module. I studied the operation manual of the modem and found, that for transferring voice via the RS232 wire, the module supports RS232-mulitplexing and wires the voice data on a separate channel (whatever this means on RS232?). Now I wonder, whether that feature is supported by miax. All what I read about, was transfering GSM voice data via bluetoth from a cell phone to miax. Does anyone succeeded in connecting a GSM-modem via miax to asterisk and transfering the GSM voice data via the RS232 cable? Thanks for any hints! Roger. P.S. Somewhere I read the advice, that I should connect the (analogue) audio connector to the PC's soundcard, which is supported by miax. But transferring the GSM voice data analogious and digitizing again afterwords is not, what I really am looking for. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
