Re: [asterisk-users] Asterisk 1.8 and dual stack support
On Sat, 23 Mar 2013 00:49:31 +, Jaap Winius wrote: There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June 2010 when I installed Debian squeeze on my server machine (while squeeze was still in its testing phase). I currently have squeeze running on almost a dozen other machines, but this file exists on only two of them. The rest all have bindv6only set to 0. I have no idea how this file got installed on these three machines; dpkg can find no record of it, so some package must have created it. The mystery of /etc/sysctl.d/bindv6only.conf has also been solved. This file was created by the postinstall script of the netbase package between v4.38 (6 Dec 2009) and v4.42 (25 Jun 2010). Apparently those earlier versions of netbase were in Debian squeeze during its testing phase (when I installed those three machines), but luckily never made it to the squeeze release version. Of course, it would have been even better if the later versions of netbase had actively changed or deleted bindv6only.conf (if present and unmodified) to remove its potential to cause problems. Cheers, Jaap -- _ -- 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] Asterisk 1.8 and dual stack support
On Fri, 22 Mar 2013 10:07:57 +0100, Jakob Hirsch wrote: This is well explained here: http://serverfault.com/a/39561 Indeed, that's the solution! There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June 2010 when I installed Debian squeeze on my server machine (while squeeze was still in its testing phase). I currently have squeeze running on almost a dozen other machines, but this file exists on only two of them. The rest all have bindv6only set to 0. I have no idea how this file got installed on these three machines; dpkg can find no record of it, so some package must have created it. However, until now it's never been a problem. I never noticed that any of the other IPv6-capable server software packages that I've been running were affected by it, until I started testing this version of Asterisk. So, why is this file here at all? It contains this comment: When IPV6_V6ONLY is enabled, daemons interested in both IPv4 and IPv6 connections must open two listening sockets. This is the default behaviour of almost all modern operating systems. If that's true, then I guess Asterisk's behavior in this case is a little out of date. But, with bindv6only set to 0, at least now when I set 'bindaddr=::' in sip.conf Asterisk will support both IPv4 and IPv6 instead of only the latter. Thanks, Jakob! Cheers, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: ... For example, if my server sends it a SIP packet with a register request and a Call-ID that looks like this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] ... somewhere along they line they end up changing it to this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1::A] Actually, I have to correct myself here. Not only was the SIP server at sip.xs4all.nl changing the lower case letters of the IPv6 section in any Call-IDs to upper case, it was also expanding the addresses, like so: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1:0:0:0:A] So, that SIP server was (and still is as of this writing) actually making two mistakes instead of just one. My apologies for not being entirely accurate the first time around. Cheers, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
On Fri, 22 Mar 2013 02:46:43 +, Jaap Winius wrote: Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) It turns out that there in a variable that can do exactly that, and is therefore the solution to this problem: 'fromdomain=domain_name'. Once placed in the [general] section of your sip.conf, Asterisk will generate Call-IDs for its SIP packets that end with an '@' followed by your chosen domain name instead of your server's IPv6 address. Thanks to Rob van der Putten for this solution! Cheers, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
On Tue, 19 Mar 2013 02:15:10 +, Jaap Winius wrote: Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider... To answer my own question, this turned out to be due to a bug in the SIP server at XS4ALL. I discovered it after using tcpdump to examine the exchange of packets during my registration attempts and noticing that Asterisk 1.8.13.1 was using an IPv6 address in the Call-ID instead of an IPv4 address as before. According to the specification for SIP 2.0 (RFC 3261) this is perfectly legal, just as long as both parties treat the entire Call-ID as a string and never make any changes to it. However, I discovered that is was exactly what the SIP server at XS4ALL is doing. For example, if my server sends it a SIP packet with a register request and a Call-ID that looks like this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] ... somewhere along they line they end up changing it to this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1::A] In other words, it is treating the latter part of the Call-ID not as a string, but as an IPv6 address and has taken it upon itself to change all of the letters in that address to upper case. This changes the Call-ID and thus my registration attemp cannot be completed. Of course, this won't affect you if you happen to have an IPv6 address without any letters in it. This situation is in contrast to another SIP provider that I use, sip.internetcalls.com, with which I currently have no problems because they leave such Call-IDs unchanged. I don't know what kind of SIP server software they use, but XS4ALL appears to be using Cirpack 4.42a. This bug is very similar to another one described in this forum exchange: http://forums.asterisk.org/viewtopic.php?f=1t=84603start=0 Here, a SIP server at an ISP was taking the IPv6 address at the end of a Call-ID and expanding it, e.g. from ::1 (the IPv6 loopback address) to 0:0:0:0:0:0:0:1. In both that case and in mine, we get the same result: an altered Call-ID that leads to endless timeouts and no registration. Hopefully, my ISP will see fit to squash this bug ASAP. -- _ -- 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] Asterisk 1.8 and dual stack support
Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only friends and peers will be able to connect to it. On the other hand, if I set it to '0.0.0.0' then it will not listen on IPv6. Is this a bug, or is this simply a limitation of Asterisk 1.8.13.1, or is there some other way to configure it for dual-stack support? Thanks, Jaap -- _ -- 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] Asterisk 1.8 and dual stack support
On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote: How are you determining that it is not listening on IPv4? bindaddr=:: should allow you to support dual stack. That's what I thought would happen. When I set bindaddr=:: and use 'netstat -lpn |grep 5060' it shows: udp6 0 0 :::5060 :::* 9898/asterisk Services like this usually also support IPv4 and as much is suggested by this comment in the sip.conf that comes with my Asterisk package: ; (Note that using bindaddr=:: will show only a single ; IPv6 socket in netstat. IPv4 is supported at the same ; time using IPv4-mapped IPv6 addresses.) However, the moment I reload my sip.conf with bindaddr=::, my entire list of IPv4-only peers loses contact with Asterisk with warnings about the network being unreachable. So, it would appear that the version of Asterisk that I'm using is operating with a single stack socket. Cheers, Jaap -- _ -- 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] Asterisk 1.8 and dual stack support
On Thu, 21 Mar 2013 16:02:17 -0700, Michael L. Young wrote: Let me try to understand this. With bindaddr set as bindaddr=::, upon starting Asterisk, you are fine and all your IPv4 peers connect properly. Therefore, dual stack is working at this point. ... You minunderstand. When I start Asterisk with bindaddr=::, the netstat output shows that it's using udp6, which usually means that the service is running in dual stack mode, but this is apparently not the case. On my system, it really is only listening on IPv6. That's why I said that, despite appearances, as soon as I reload SIP (or restart Asterisk) with this setting, I lose contact with my entire list of IPv4-only peers, while Asterisk gives warnings about the network being unreachable (the IPv4 network). I've also tried using multiple bindaddr lines with a mix of IPv4 and IPv6 addresses, but then the service ends up binding only to the last address. Therefore, it looks to me like the version of Asterisk that I'm running is only capable of running in single stack mode, supporting either IPv4 or IPv6, but not both at the same time. Upon issuing a sip reload, your peers lose their ability to communicate with Asterisk? Is that correct? That's right. What does netstat -lpn |grep 5060 show after the reload? udp6 0 0 :::5060 :::* 9898/asterisk These network unreachable warnings are from Asterisk or your peers? From Asterisk. They look like this for two of my IPv4 SIP devices: [Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer '1000' is now UNREACHABLE! Last qualify: 110 [Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer 'patton' is now UNREACHABLE! Last qualify: 20 I also get errors for connections to SIP servers for which I have register entries in the [general] section of sip.conf. The errors for one of them, sip.xs4all.nl, which is IPv4 only, look like this: [Mar 21 23:24:14] ERROR[9931]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo(sip.xs4all.nl, (null), ...): No address associated with hostname [Mar 21 23:24:14] WARNING[9931]: acl.c:582 resolve_first: Unable to lookup 'sip.xs4all.nl' Anyway, as soon as I reload sip without bindaddr=::, these errors stop. What version of Asterisk are you using? Version 1.8.13.1. Asterisk 1.8.0 had IPv6 support in it. Therefore, every minor version released since would still have IPv6 support in it. That's good to know, so maybe it's just my minor version that has a bug that prevents it from running in dual stack mode. That's what my question was about in the first place. Cheers, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote: Hopefully, my ISP will see fit to squash this bug ASAP. Well, I got my answer from them quickly enough: Nope. Luckily, somebody was kind enough to suggest a workaround. Unfortunately, it involves, downloading the source code and making a few changes to it to prevent Asterisk from adding '@IPaddress' to the end of the Call-ID string. Nevertheless, it's easy enough to do. The idea is to look for this string that appears twice in ./channels/chan_sip.c: ast_string_field_build(pvt, callid, %s@%s, generate_random_string(buf, sizeof(buf)), host); And to change it to: ast_string_field_build(pvt, callid, %s, generate_random_string(buf, sizeof(buf))); Now my Call-IDs look like this: Call-ID: 63935a8d2144d4f1309024fd7612f608 Instead of this: Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a] Still, I'd much prefer that my ISP fixed the problem instead, because now every time a security update becomes available for Asterisk, I'm going to have to download the source code, make the same changes, recompile it and install it all over again and again. Ho hum. Of course, an even better solution would be if Asterisk had a variable with which to alter the Call-ID string format so that I could omit the IP address. :-) Cheers, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
On Tue, 19 Mar 2013 11:58:22 +0100, Asghar Mohammad wrote: try srvlookup=yes Already tried that, but enabling DNS lookups makes no difference when establishing the SIP connection. The error message that I keep seeing at the console looks like this: [Mar 19 12:47:21] NOTICE[7494]: chan_sip.c:13171 sip_reg_timeout:-- Registration for 'telno@sip.xs4all.nl' timed out, trying again (Attempt #3) Incidentally, I have remote access to two other Asterisk systems in the Netherlands with XS4ALL connections, both still Debian squeeze with Asterisk 1.6.2.9, and when I add my register line to their sip.conf files, which are virtually identical to mine (except for the context), it registers immediately. This shows that my account still works. Moreover, I also have remote access to some more Asterisk systems with XS4ALL connections and Debian wheezy with Asterisk 1.8.13.1. When I add my register line to their sip.conf files, which are virtually identical (except for the context), it fails. For the rest those systems are pretty much like my own, but at least it demonstrates that the problem is not unique to my system and connection. Oh, and all of these systems have srvlookup=no (default is yes). Thanks anyway, Jaap -- _ -- 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] SIP account registration fails after upgrade to 1.8
Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my sip.conf looks like this: [general] context=incoming-j canreinvite=no dtmfmode=inband qualify=yes srvlookup=no disallow=all allow=alaw allow=ulaw allow=g722 allow=g726 allow=g729 insecure=port,invite register = telno:password@sip.xs4all.nl/telno Does anyone know of any new variables that have been introduced since Asterisk 1.6.2.9, that apply here and might be causing this problem? Thanks, Jaap -- _ -- 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] Forcing a CODEC
Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- 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] IAX2 availability testing
Hi folks, What methods are available for testing IAX2 service availability? I know about iax2 show peers and iax2 show registry, but I'd like some alternatives. Tcpdump shows a little more about what's going on, but a handy test using nmap doesn't seem to work anymore (see http://shearer.org/UDP_Reachability_Testing). Any suggestions would be appreciated. Cheers, Jaap PS -- My systems run Debian squeeze with Asterisk 1.6.2.9. -- _ -- 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] Logging the CID from the Privacy Manager
Quoting Matt Riddell li...@venturevoip.com:
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
That looks so simple -- and it actually works! -- although exactly not
in the way that I was expecting. Instead of replacing the contents of
one of the existing fields, a new field, userfield, appeared at the
end of the record containing the number submitted by the caller.
I did try to use the same method to change one of the existing fields,
e.g. src, like this:
Set(CDR(src)=${CALLERID(num)})
But, then I received this error:
[Sep 1 12:26:15] ERROR[12562]: cdr.c:303 ast_cdr_setvar:
Attempt to set the 'src' read-only variable!.
That doesn't seem to be possible. So, I'm happy with your solution.
Thanks, Matt!
Cheers,
Jaap
This message was sent using IMP, the Internet Messaging Program.
--
_
-- 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] Logging the CID from the Privacy Manager
Hi folks,
My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL
database, including those handled by the Privacy Manager.
Unfortunately, even though I can use the CLI to see the information
being submitted by anonymous callers to satisfy the demands of the the
Privacy Manager, that information is not recorded in the database.
Instead, all that is written to it:
clid: Privacy Manager anonymous
src:anonymous
Can the number submitted to the Privacy Manager somehow be recorded in
the database, instead of anonymous?
Thanks,
Jaap
PS -- Currently, the configuration I'm using in the dialplan for the
Privacy Manager looks like this:
exten = jw,1,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
exten = jw,n(bad),Playback(im-sorry)
exten = jw,n,Playback(vm-goodbye)
exten = jw,n,Hangup()
This message was sent using IMP, the Internet Messaging Program.
--
_
-- 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] Asterisk 1.6 and PrivacyManager with SIP
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} = anonymous, instead of being blank. So, to get it
to work I added the first three rules to the following:
exten = jaap,1,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jaap,n(true),Set(CALLERID(num)=)
exten = jaap,n(false),NoOp()
exten = jaap,n,PrivacyManager()
exten = jaap,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jaap,n,Dial(SIP/1000,20,w)
exten = jaap,n,Hangup()
exten = jaap,n(bad),Playback(im-sorry)
exten = jaap,n,Playback(vm-goodbye)
exten = jaap,n,Hangup()
Unfortunately, this no longer seems to work with Asterisk 1.6: the
second rule is still executed, but for some reason the PrivacyManager
always decides that the caller ID is present anyway.
Should I be doing this differently now, or is something else wrong?
Thanks,
Jaap
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and PrivacyManager with SIP
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before Asterisk
even reaches that rule, the CLI shows that the result from the
PrivacyManager is:
-- CallerID Present: Skipping
PrivacyManager is simply failing to determine that the incoming SIP
calls are anonymous.
Actually, could it be that the second rule of my code, with the Set()
command, is simply not working with Asterisk 1.6? Let me try that
without the empty set of quotes after the equals sign...
Yes, that was it -- it's working again! Here's what it looks like now:
exten = jaap,1,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jaap,n(true),Set(CALLERID(num)=)
exten = jaap,n(false),NoOp()
exten = jaap,n,PrivacyManager(3,10)
exten = jaap,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jaap,n,Dial(SIP/1000,20,w)
exten = jaap,n,Hangup()
exten = jaap,n(bad),Playback(im-sorry)
exten = jaap,n,Playback(vm-goodbye)
exten = jaap,n,Hangup()
Rule five now has both ${PRIVACYMGRSTATUS} and FAILED without quotes,
but that actually did not make any difference. Two things actually
fixed the problem. The first and most important was removing the pair
of empty quotes from rule two -- otherwise the caller ID is no longer
regarded as empty. Second is the addition of 3,10 as options to the
PrivacyManager application in rule four. Those are supposed to be the
defaults, but without them the PrivacyManager fails to recognize a
ten-digit phone number as being sufficient. I consider that a bug.
Cheers,
Jaap
--
_
-- 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] ISDN config: LBO values
Quoting Tilghman Lesher tles...@digium.com: http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php See pages 17-18 of the associated PDF. While this is not the T1 framer chip used, the values are identical, which leads me to believe that these values are actually industry standard. Well, maybe more like a defacto standard. But, it still doesn't explain when to use the different values in a software configuration, e.g. with Asterisk. As a term, DSX-1 is confusing. One description can be found in the Wikipedia article for T-carrier, which says it stands for Digital Signal Crossconnect: DS1 signals are interconnected typically at Central Office locations at a common metallic cross-connect point known as a DSX-1. ... On the other hand, articles like the following use DSX-1 to describe customer site connections: * Adtran NetVanta T1 Access Router http://www.arcelect.com/netvanta_access_t1_router.htm The diagram shows how two different NetVanta models can be used to connect a T-1 line to a PBX. There's also this page: * Primary Rate Interface ISDN Line Port http://www22.verizon.com/wholesale/solutions/solution/pri+rate+isdn.html Near the end, under Detailed Information, it says: PRI service consists of a 4-wire DSX-1 port associated with a local switching system and the 4-wire DSX-1 cross-connect between the OTC DSX-1 termination and the local switching system DSX-1 termination. PRI ports are DSX-1 interfaces that meet the electrical specifications in ANSI T1.102. PRI service and use B8ZS line code and the Extended Superframe Format (ESF) described in ANSI T1.403. Again, the term DSX-1 is used to describe a CPE port. In such cases, I think it will probably be appropriate to use the DSX-1 column in the LBO table. Still, what's the difference between CSU and DSX-1?? Speculation: Could it be that CSU refers to situations where there is no equipment of any kind between the demarcation point and the ISDN card? In such cases, the ISDN card will have an integrated CSU, and the length of the cable will be unknown (thousands of feet), but you can know the attenuation value in dB; either by measuring it, or by getting it from the telco. This scenario may only occur in the United States. On the other hand, DSX-1 will refer to situations where the ISDN card is connected -- via a DSX-1 port and a cable of a known length -- to an external CSU and/or DSU. In turn, this equipment is connected to the demarc. This scenario may apply in all other situations, e.g. ISDN BRI cards that connect to an NT-1. Does this sound reasonable? Thanks, Jaap -- _ -- 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] ISDN config: LBO values
Quoting Tilghman Lesher tles...@digium.com: The value selected should almost always be zero. However, if the cable is of a significant length, another value must be selected, but which one? There are two columns: CSU and DSX-1. When is it appropriate to use the one or the other to determine the correct LBO value? Each LBO value is a different amount of loss to be expected on the line, and therefore the signal is amplified a commensurate amount. What it really comes down to is what works for you. That's the usual approach, but if I was still happy with it I would not have asked the question. According to the manual, the values are found in a table, but what good is that if you can't make any sense of it? In the mean time, I've googled some more and found one text that suggests CSU and DSX-1 are both T1 trunk interface types, while another suggests that a DSX-1 is an interface that a CSU is attached to. It seems to me that the table refers to two situations that used to (or maybe still do) occur in North America in which an ISDN card is be attached to a T1 trunk line via a CSU/DSU (the DSX-1), or only a CSU. In the latter case, the ISDN card must also act as a DSU. Can anyone say is this is correct? Any further explanation would be welcome. Cheers, Jaap -- _ -- 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] ISDN config: LBO values
Hi all, When configuring Asterisk with an ISDN card, it will at one point become necessary to select the LBO (Line Build-Out) value. This is an integer (0-7) that is determined by the length of the cable and is selected from the following table. Many of us are familiar with it: CSU (dB) DSX-1 (feet) --- 00 0?133 1 133?266 2 266?399 3 399?533 4 533?655 5-7.5 6-15 7-22.5 The value selected should almost always be zero. However, if the cable is of a significant length, another value must be selected, but which one? There are two columns: CSU and DSX-1. When is it appropriate to use the one or the other to determine the correct LBO value? Thanks, Jaap -- _ -- 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] Simulating a commercial SIP provider
Quoting Alfredo Peña arp...@gmail.com: Try using this line in the [general] section of sip.conf in your simulated SIP provider machine: realm=sip.provider.com No, that didn't seem to make any difference. However, this did: insecure=invite This prevents the Failed to authenticate on INVITE errors from occurring on both sides when INVITE messages arrive with user names (before the @ sign) that are only known on the remote system. The user names are associated with the phones that I use on either end of the connection. Unfortunately, I'm forced to use this option on both sides of the connection, instead of only on the provider side. Therefore, it's still not really the answer that I'm looking for, but it's a step in the right direction. Cheers, Jaap -- _ -- 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] Simulating a commercial SIP provider
Hi all, The kind of configuration that I use in my sip.conf to connect to various commercial SIP providers looks like this: [general] context=incoming-calls canreinvite=no qualify=yes register = jwinius:pass...@sip.provider.com/0201234567 [provider] type=peer host=sip.provider.com fromuser=jwinius secret=passwrd This works. However, how would I have to configure the sip.conf of a second Asterisk machine if I wanted to use it to simulate the host mentioned above, sip.provider.com, but (crucially) without changing the above configuration? I would have thought that the appropriate stanza to use for my account in the other Asterisk machine's sip.conf -- the system that simulates the commercial SIP provider -- would have to look like this: [jwinius] type=friend host=dynamic secret=passwrd insecure=invite Unfortunately, this doesn't work, resulting Failed to authenticate on INVITE errors. It only works if I first remove the fromuser and secret options from the configuration on the first system, but that's not what I want. Any idea what I'm doing wrong? Thanks, Jaap -- _ -- 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] Simulating a commercial SIP provider
Quoting Motiejus Jak?tys desired@gmail.com: If I understand well - you want second PBX to act as your sip.provider.com add this to your /etc/hosts (on primary pbx): 10.10.10.10 sip.provider.com No, I'm afraid you misunderstand. This has nothing to do with DNS and not being able to reach my second PBX -- that's all fine. The hostname, sip.provider.com, is fictitious anyway. The problem is how to configure the client entry in the second PBX's sip.conf so that the first PBX can use it without having to change anything (other than the hostname). As things stand, I can already do it, but only if I first remove the fromuser and secret options from the sip.conf of the first PBX: that's going too far. Eventually, I hope to use the new information to expand this article: * Asterisk: minimal SIP configuration http://www.rjsystems.nl/en/2100-asterisk.php The text would start with: If a second Asterisk server is used to simulate the connection to the commercial SIP provider, add this stanza to its sip.conf ... Thanks anyway, Jaap -- _ -- 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] cat /proc/zaptel/*
Quoting Jaap Winius jwin...@umrk.to: Being both impatient and charitable, I'll try answering this myself: ISDN uses LAPD for the D-channel and LAPB for data connections over the B-channels. However, LAPB is irrelevant for Asterisk, because when the B-channels are used for voice they carry no signaling. This is why it is necessary to specify a line code protocol, such as AMI, for the B-channels, and a frame type, typically CCS, for the D-channel. Would that statement be correct? Basically, although the last line is a little muddled. Also, would someone care to elaborate on how the CCS protocol fits into this picture, in particular how it relates to LAPD? LAPD, described in ITU-T recommendations Q.920 and Q.921, is an OSI network layer 2 protocol, while CCS (Common Channel Signaling), which is described by Q.930 (I.450) and Q.931 (I.451), is layer 3. Two things that I found confusing here are: 1.) The documentation that explains the Zaptel span configuration statement (in /etc/zaptel.conf) describes the D-channel signaling type as framing, which I find misleading. IMO signaling would have been more accurate. 2.) CCS is a connection control signaling type. The problem is that there is more than one CCS type, although my impression is that the one used most often for the ISDN D-channel is Q.930/Q.931. The others I've heard of are QSIG CCS (Q.931/Q.933) and SS7 (Q.700-series with many variants). Cheers, Jaap -- _ -- 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] cat /proc/zaptel/*
Hi all, Thanks to Russ Meyerriecks for his previous reply in this thread, which was very informative. I'm now hoping that someone will comment on the following: ISDN uses LAPD for the D-channel and LAPB for data connections over the B-channels. However, LAPB is irrelevant for Asterisk, because when the B-channels are used for voice they carry no signaling. This is why it is necessary to specify a line code protocol, such as AMI, for the B-channels, and a frame type, typically CCS, for the D-channel. Would that statement be correct? Also, would someone care to elaborate on how the CCS protocol fits into this picture, in particular how it relates to LAPD? Thanks, Jaap -- _ -- 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] cat /proc/zaptel/*
Hi all, On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is: ~# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A Zaptel Driver card 1 [TE] AMI/CCS 4 ZTHFC2/0/1 Clear 5 ZTHFC2/0/2 Clear 6 ZTHFC2/0/3 HDLCFCS These are two HFC-S PCI A cards. But, what exactly does all of this mean? In particular: * Span - In telephony, what is the definition of this term? * MASTER - How is this relevant? Only for timing purposes? * Clear - Is this said because only B-channels use ISDN clear codes? * HDLCFCS - Why say this about D-channels? Why not just say HDLC? * (In use) - What does this mean and how is this state determined? * 1 ZTHFC1/0/1 Clear (In use) - What do each of these columns specify? Thanks, Jaap -- _ -- 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] Problem with Sangoma A104 and euroisdn pri
Quoting RESEARCH resea...@businesstz.com: Can you post outputs for the following commands; #asterisk -rx 'pri show spans' #asterisk -rx 'zap show channels' #wanpipemon -i w1g1 -c Ta Sure thing! Here they are in succession: == # asterisk -rx 'pri show spans' PRI span 1/0: Provisioned, Down, Active # asterisk -rx 'zap show channels' Chan Extension Context Language MOH Interpret pseudodefaultdefault 1defaultdefault 2defaultdefault 3defaultdefault 4defaultdefault 5defaultdefault 6defaultdefault 7defaultdefault 8defaultdefault 9defaultdefault 10defaultdefault 11defaultdefault 12defaultdefault 13defaultdefault 14defaultdefault 15defaultdefault 17defaultdefault 18defaultdefault 19defaultdefault 20defaultdefault 21defaultdefault 22defaultdefault 23defaultdefault 24defaultdefault 25defaultdefault 26defaultdefault 27defaultdefault 28defaultdefault 29defaultdefault 30defaultdefault 31defaultdefault # wanpipemon -i w1g1 -c Ta * w1g1: E1 Alarms (Framer) * ALOS:OFF| LOS:OFF RED:OFF| AIS:OFF OOF:OFF| RAI:OFF * w1g1: E1 Alarms (LIU) * Short Circuit:OFF Open Circuit:OFF Loss of Signal:OFF * w1g1: E1 Performance Monitoring Counters * Line Code Violation: 324 Far End Block Errors: 0 CRC4 Errors: 0 FAS Errors: 0 Rx Level: -2.5db # == Cheers, Jaap -- _ -- 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] Problem with Sangoma A104 and euroisdn pri - SOLVED
Quoting James Lamanna jlama...@gmail.com: I would call KPN Telecom and ask them for help as well. They will have much more sophisticated tools for debugging PRIs and also will be able to check on their end if they see the D-Channel as up. After studying the configuration more closely, the first thing I changed was... === begin wanpipe1.conf === [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 13 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL= 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF= NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES === end wanpipe1.conf = This was my last result after running /usr/sbin/wancfg, although I'm not sure the changes made any difference. Probably minor. The second step was to modify /etc/zaptel.conf to reflect the wanpipe1.conf configuration: === begin /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,0,ccs,hdb3 bchan=1-15,17-31 hardhdlc=16 === end /etc/zaptel.conf == The span line used to be span=1,0,0,ccs,hdb3. The second field specifies the timing source, which in my case needs to be the provider (KPN), so I changed it to a 1. I also took the opportunity to changes the loadzone and defaultzone to nl. However, I noticed later that the old configuration works as well, so these changes were evidently not too important. In the end, the real problem turned out to be on the other end of the line. After opening a trouble ticket with KPN Telecom, at one point the pri just started to work: = CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 = Also, this is what now appears at the CLI shortly after starting Asterisk: = -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 -- B-channel 0/24 successfully restarted on span 1 -- B-channel 0/25 successfully restarted on span 1 -- B-channel 0/26 successfully restarted on span 1 -- B-channel 0/27 successfully restarted on span 1 -- B-channel 0/28 successfully restarted on span 1 -- B-channel 0/29 successfully restarted on span 1 -- B-channel 0/30 successfully restarted on span 1 -- B-channel 0/31 successfully restarted on span 1 = I guess that's good. When I asked KPN what they had done, they said they hadn't discovered any problems. However, I was also told that the line was reset as a normal part of their troubleshooting procedure, so I guess that made the difference. Problem solved. Thanks! Jaap -- _ -- 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] Problem with Sangoma A104 and euroisdn pri
Hi all,
My problem boils down to these errors:
... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time
This is triggered by lines in extentions.conf such as:
exten = _X.,1,Dial(ZAP/g1/${EXTEN},,W)
The system is CentOS v5.2 with Asterisk 1.4.23
(druid-asterisk-1.4.23.1-2), a Sangoma A104 4-port card, Wanpipe
v3.4.4 and Zaptel v1.4.12.1. The system is attached to a single
EuroISDN PRI and is located in the Netherlands.
Besides the above error, I also noticed this:
CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3
The status needs to be Provisioned, Up, Active.
Following Sangoma's instructions for debugging an Asterisk PRI span, I
can confirm that there are only outgoing frames and that the D-channel
messages in Asterisk are the same as what the Wanpipe drivers are
seeing. So, assuming that my local telco (KPN Telecom) has activated
the D-channel, what else could possibly be causing this problem?
Thanks,
Jaap
PS -- Below are my current configuration files and debugging output:
==begin zaptel.conf
loadzone=us
defaultzone=us
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
hardhdlc=16
==end zaptel.conf ==
==begin wanpipe1.conf ==
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment
[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS = 13
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE= 1
TE_CLOCK = NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE= NO
MTU = 1500
UDPPORT = 9000
TTL= 255
IGNORE_FRONT_END = NO
TDMV_SPAN= 1
TDMV_DCHAN= 16
TDMV_HW_DTMF= NO
TDMV_HW_FAX_DETECT = NO
[w1g1]
ACTIVE_CH= ALL
TDMV_HWEC= NO
==end wanpipe1.conf
==begin zapata.conf
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
switchtype=euroisdn
context=default
group=1
signalling=pri_cpe
channel =1-15,17-31
==end zapata.conf ==
Here's some debugging output:
=== begin debug info ==
# ztcfg -vv
Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Hardware assisted D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)
31 channels to configure.
# wanrouter status
Devices currently active:
wanpipe1
Wanpipe Config:
Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK |
Baud rate |
wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 169 | 4 | 1
| N/A |
Re: [asterisk-users] SIP source address error -- fixed
Quoting Jaap Winius jwin...@umrk.to: The question remains: how can a remote Asterisk server be receiving SIP packets that still contain the private net IP address of a client? Okay, I fixed it: by installing siproxd on the firewall system of the local network. With the Debian systems I'm running, I let iptables take care of NAT. Last December, with kernel 2.6.24, I didn't need a SIP proxy to get a SIP client to register with a remote Asterisk server. Now, with 2.6.26, I do. Conclusion: NAT sucks. If we were all using IPv6, this would not be an issue. Jaap ___ -- 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 source address error
Quoting Matt Riddell li...@venturevoip.com: [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1: Operation not permitted Are you binding to an address that the box doesn't own? Check the top of sip.conf. It's set to bind to 0.0.0.0, which IIRC is nothing strange. The question remains: how can a remote Asterisk server be receiving SIP packets that still contain the private net IP address of a client? Jaap ___ -- 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] SIP source address error
Hi all, My Asterisk problem today involves getting a SIP client on a private net to register with a server somewhere else on the Internet. This worked for me about a year ago no problem, but now I see an error message on the remote server every time the client attempts to connect (the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3). Here's an example: [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1: Operation not permitted 192.168.8.30? At first I thought maybe the local NAT (iptables SNAT) wasn't doing its job properly, but it seems fine for the rest. Also, the same client, going through the same NAT, has no problem connecting to my ISP's SIP server. Then I thought it might be the SIP client (a Siemens Gigaset S675IP phone), but I get exactly the same problem when using an old analog phone with a Linksys SPA-3000 instead. Has anyone encountered this problem before? If so, what caused it and what solved it? Thanks, Jaap ___ -- 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] Best ISDN BRI solutions?
Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be? I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ISDN BRI solutions?
Quoting Jorge Mendoza mend...@tcc.com.pe: We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. What a great idea! I'm going to remember that. Unfortunately, I believe that would be of no use if you also wanted to use your ISDN connection for a networked fax system, such as with Hylafax and IAXmodem. Cheers, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debug: how to print a variable?
Hi all,
Is it possible to display or print variables in Asterisk (e.g. in the
CLI) for debugging purposes?
For example, I'm using two different types of SIP phones: the Snom M3
and the Siemens S675IP. However, when anonymous callers submit a
number to the PrivacyManager, only the Siemens displays the new CID
correctly; the Snom shows unknown (even though the new CID looks
okay in the database). That's as opposed to when the CID is visible to
begin with, in which case both phones display the CID correctly.
For this reason it seems to me that there is a difference between a
normal CID and the one generated by the PrivacyManager. If I knew what
that difference was, perhaps I could correct it with some more
scripting, but I don't know of any way to display such variables. In
this case I believe the relevant variable to display is
${CALLERID(num)}.
Can anyone help?
Thanks,
Jaap
___
-- 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] PrivacyManager no longer working properly
Quoting Jaap Winius jwin...@umrk.to: Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten = jaap,n,PrivacyManager(1,1) ... (I'm not using a privacy.conf file), the submitted caller ID is always considered invalid. This issue has been resolved, at least on my system. After running some more tests, I discovered that the PrivacyManager was only having problems with calls coming in via SIP; anonymous calls incoming via ISDN were treated normally. The Asterisk version I was using was from Xorcom (1.4.21.2~dfsg-3 for Debian lenny). Thinking that the version might be a problem, I first decided to try for an upgrade. I noticed that Xorcom had a major update in store for me -- Asterisk v1.6.1.0~dfsg-1 -- but worried that the corresponding replacement of zaptel with dahdi software would cause problems (I need it to support my HFC-PCI card). Nevertheless, I gave it a try. Bad idea. I wasted several hours late last night trying to get the HFC-PCI card working working with dahdi, but without any luck. The first thing I noticed was that the zaphfc module was still there (not renamed), while the one that I prefer -- vzaphfc -- was not. To get dahdi_genconf to work I found that it was important for dahdi_dummy be loaded after zaphfc. That went fine, but then running dahdi_genconf would lock up the system, with thousands of error messages flashing across the server console: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. After a few of these lock-ups and reboots, I abandoned the upgrade. Obviously, I'll try for it again at a later date, but I really do hope that by that time I will discover that dahdi includes a working equivalent of vzaphfc. Not wanting to go against the grain by attempting to manually reinstall and then freezing the older asterisk and zaptel packages from Xorcom, which would certainly get me nowhere as far as my privacymanager problem was concerned, I decided at this point to try to install the stock version that comes with Debian lenny instead. After installing all of the necessary packages, I saw that the HFC-PCI card was working again, but so was the privacymanager (for both ISDN and SIP). All of my problems were solved! In hindsight, however, I see that I've been running the stock Debian versions of Asterisk and Zaptel for lenny all along. I was running v1.4.21.2~dfsg-3 before, just as I am now, but since Xorcom was until recently only offering an older version for Debian lenny, 1.4.21.1~dfsg-0.5941, apt wasn't selecting it. The same can be said for the Zaptel packages that I have installed now compared to before (1.4.11~dfsg-3), except that before I also had an even older zaptel-firmware package installed, 1.4.10.1-0.567, which must have come from Xorcom. I don't think that it was influencing matters, though, since the compiled zaptel-modules packages are still the same version now as before. So, how come the privacymanager is working 100% now? No idea. Thanks to my fantastic backup system, I'm also using the same Asterisk configuration files now as I was before. It's a mystery I guess. In the mean time, I will see if I can acquire an extra HFC-PCI card from somewhere and set up a new system with which to test Asterisk 1.6. Cheers, Jaap ___ -- 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] PrivacyManager no longer working properly
Hi all, Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten = jaap,n,PrivacyManager(1,1) ... (I'm not using a privacy.conf file), the submitted caller ID is always considered invalid. Does anyone recognize this problem? Does the PrivacyManager have a new parameter that I'm missing? Thanks, Jaap ___ -- 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] Reliable wireless SIP phones
Hi list, Are there any reliable wireless SIP phones available on the market yet? Six months ago I went for the Siemens Gigaset 675IP. Although there was a bug in the MWI support, unit #1 seemed fine for the first few weeks, so I bought #2 and #3. Then the problems started. Of the three units, two regularly lock up, which requires the base station to be reset, and two do not perform DTMF menu navigation properly (I live in the Netherlands). There was a recent firmware update for this model, but there may as well not have been. You'd think that a big company like Siemens that has been making rock-solid DECT phones (analog and ISDN) for years could do better, but apparently not. Since the firmware seems to be the same, there's no way I'm going to upgrade to the 685IP. I was thinking of trying out the Snom M3, but according to voip-info.org, that model suffers from similar reliability problems. There is also the possibility of using a Wi-Fi SIP phone instead, but I haven't been able to find a positive review of one of these phones either, despite the promising concept. So, what's the most reliable wireless SIP phone these days: an analog DECT phone with a Linksys SPA adapter? Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch Asterisk mailing list?
Quoting Erik de Wild: Tripple-o [EMAIL PROTECTED]: What is the most reliable method for Asterisk to detect the Called ID for incoming calls on an analog line in the Netherlands? In Holland you have to pay to receive cid info on the incoming line. I've got that and I've tested it (twice), so I know it works. ... if you have a phone that only supports FSK the CID will never work. I've got one of those too; I used it for the above test. I still have a couple of ETSI - FSK converters catching dust. So if you pay for CID but your phone doesn't support and you have a pot line connected to your Asterisk server I can provide you with a solution for a couple of EUR. Aha! That's definitely a workaround, but it sounds like it should work. If you use the proper card maybe you can adjust the settings so it supports ETSI instead of FSK. The proper card? Which one would that be? I've been told that my TDM410 should work, but as I've said before, no luck so far. I'm starting to believe that it may involve a magical combination of settings along with chicken blood, frog entrails and some sort of dance (a solution more familiar to Windows admins). I used X100P cards and needed the convertor to get proper CID. Tired of wasting time, I suspect this will be the quickest way out for me as well. If the Dutch mailing list starts I will join ;-) I just added myself to the new list. Cheers, Jaap ___ -- 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] Dutch Asterisk mailing list?
Hi folks, Would anyone here happen to know of the existence of a Dutch Asterisk mailing list? If so, where can it be found? It's not that I'm unable to pose my questions here in English, but I'm hoping that I may sooner find an answer there to the following question: What is the most reliable method for Asterisk to detect the Called ID for incoming calls on an analog line in the Netherlands? So far, I've tried using a Linksys SPA3000 and an SPA3102, as well as a Digium Wildcard TDM401BF for this purpose, but all to no avail. I suspect that there is a solution, but perhaps the people who are familiar with it like to hang out somewhere else. At least, I hope that's the case. Can anyone help? Thanks Jaap ___ -- 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] Digium TDM4xx CID problem
Hi list, Has anyone here used one of these cards and got it to recognize incoming CIDs in Denmark, Sweden, or the Netherlands? I'm still looking for a way to attach an analog line to my Asterisk system in the Netherlands that recognizes incoming CIDs. I've now purchased a Digium Wildcard TDM401BF: a basic card with a single FXO module and no echo cancellation. Yes, it's more expensive than a Linksys SPA3102, but if anything was supposed to work it was this, but so far it ain't. Most recently, I was pointed to this bug report: http://bugs.digium.com/print_bug_page.php?bug_id=9 Looks very relevant! However, even though it seems to apply to Asterisk v1.4.19 (the version I'm currently using) this issue was closed almost four years ago and it looks like the diff files have long since been added to the main development tree. Anyway, except for not detecting incoming CIDs, the card works fine. My zapata.conf looks like this: [trunkgroups] [channels] language=en rxwink=300 cidsignalling=dtmf relaxdtmf=yes cidstart=polarity usecallerid=yes callerid=asreceived callwaitingcallerid=yes callwaiting=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=14 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming group=1 signalling=fxs_ks callerid=asreceived context=from-pstn channel = 1 Any ideas? Thanks, Jaap ___ -- 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] Siemens Gigaset S685IP Review
Quoting Marco [EMAIL PROTECTED]: * The firmware and ALL of the pre-recorded messages are in german. I had some customers a little scared about this! I have German units too (of the S675IP), but it's easy to switch the menu language to English. If the pre-recorded messages are still in German (and I believe they are), then I don't think anyone has noticed, since I let Asterisk do my voicemail for me. However, if you want to use the unit's integrated answering machine -- in English -- I would think that would just be a question of finding and installing a UK firmware version... unless maybe the pre-recorded messages are separate files. Cheers, Jaap PS -- I opted for the German versions because they cost significantly less. ___ -- 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] Roaming callback?
Quoting Jerry Harshany [EMAIL PROTECTED]: There is an additional alternative for a ringback to a caller, which is to use the Call File capability as noted in Van Meggelen's Future of Telephone; 2nd ed, p306. As it says in the book, call files allow calls to be created through the Linux shell. If you've used this to create a roaming callback service, then you must have created something that allows users to submit a phone number to be called back on, after which a .call file is created and moved to the /var/spool/asterisk/outgoing/ directory. sleep 8s mv $1 $2 exit 0 This looks like the step that moves the newly created call file to the aforementioned directory. In my case, when the caller calls in to 'asterisk', he is prompted for the number he wishes to call. The caller can be at a US or international number, and he can call any US or international number, WITH or WITHOUT ringback. In other words the caller designates whether this is a direct connect call, or a ringback (and then bridge the called number). I have the complete flexibility of my dial plan extensions to do as I wish with the phone numbers. This is what I'm really interested in! How did you manage this? Would you be willing to share how you did this? Cheers, Jaap ___ -- 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] Siemens Gigaset S685IP Review
Quoting Michael Graves [EMAIL PROTECTED]: in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. ... Sounds great, especially where you say that you got MWI to work with Asterisk. I bought a couple of S675IP units a few months ago and have never been able to get MWI to work. The relevant lines that I've added to my phone's SIP config are: subscribemwi=yes [EMAIL PROTECTED],1234 I'm using Asterisk v1.4.19. AFAIK, this is a firmware problem that, according to voip-info.org, also affects the c470ip (same firmware), but Siemens has yet to fix. Cheers, Jaap ___ -- 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] Roaming callback?
Hi list, Regarding callback functionality, it seems that Asterisk only includes a provision for callback in the voicemail configuration, for authorization purposes, but not an actual callback mechanism. For that, there are various free 3rd party AGI (Asterisk Gateway Interface) scripts available: * Asterisk tips callback http://www.voip-info.org/wiki/view/Asterisk+tips+callback * capi Callback http://www.junghanns.net/en/callback.html Looking at the scripts, they don't seem too difficult to implement, but they don't exactly work as I was hoping either. First, they require your system to have a dedicated callback number that you ring once and then hang up. The system then calls you back at a predefined number, e.g. your mobile phone. Not a very flexible solution. What I had in mind was an option in the voicemail menu that would allow you to dial a number -- any number -- at which the system would call you back. I'd call this roaming callback. Is anything like this available for Asterisk? Cheers, Jaap ___ -- 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] PSTN gateway alternatives
Hi list, The Linksys SPA-3000 and SPA-3102 are often used as PSTN gateways for Asterisk. They're cheap and convenient to use. Both have worked fine for me, except I've never been able them to pass on incoming Caller IDs. I know about the PSTN CID For VoIP CID and Caller ID Method settings and use the most recent firmware versions, but it makes no difference. Perhaps this Linksys functionality just doesn't work in the Netherlands, where I live. Can anyone suggest an alternative to these devices, particularly something that is known to be more reliable at passing on incoming CIDs? Thanks, Jaap ___ -- 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] PrivacyManager not working
Hi list, On my system, PrivacyManager is not reacting to anonymous calls. Whenever I dial into my system with my mobile phone's number hidden, the CLI message CallerID Present: Skipping shows up and and my SIP phone rings anyway. Perhaps the cause is due to the fact that when there is no CID, the results are not always the same. For example, I see in my CDR database that when anonymous calls come in via the SIP channel, the clid field shows: Anonymous anonymous However, when anonymous calls came in though my old ISDN line (which I can't test anymore because it no longer exists), the clid field would show: CID withheld Although I'm not sure, I suspect that PrivacyManager recognizes the latter format, but not the former. Has anyone else experienced this problem, or know of a fix or workaround? FYI: I'm using Asterisk 1.4.19 and anonymous calls only come in via SIP. Thanks! Jaap ___ -- 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] Linksys SPA devices and CID
Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA device, all I see is the SPA device CID associated with the PSTN line; not the CID of the incoming call. The only SPA settings I know of that are supposed to enable the passing on of PSTN CIDs are the PSTN CID For VoIP CID option (under PSTN Line), which AFAIK must be set to yes, and the Caller ID Method (under Regional), which I must set to ETSI DTMI With PR, or else my analog phone will not display any CIDs when attached to the SPA's FXS port. Yet, these settings have never led to any positive results, despite attempts with different firmware versions on both devices. Can anyone help? Thanks, Jaap ___ -- 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] Linksys SPA devices and CID
Quoting Tim Johnson [EMAIL PROTECTED]: Your caller ID is probably being over-ridden by the settings in your sip.conf file. Remove the caller ID from your PSTN section of the sip.conf, and the CID should be passed on from the POTS line. That sounds like a good idea regardless. On the SPA3000 I've changed the User ID to PSTN, while the sip.conf now has the following entry: [4500] ; SPA3000, PSTN line: incoming. type=friend host=dynamic port=5061 context=home-in username=PSTN secret=1234 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very qualify=yes While still not a solution in my case, this is an improvement. CIDs for incoming PSTN calls are now reported as Unavailable, instead of always being 4500. Thanks! Jaap ___ -- 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] Linksys SPA devices and CID
Quoting Tim Johnson [EMAIL PROTECTED]: What do you have for your PSTN Answer Delay (in PSTN tab)? I had to set mine between 3 to 5 to get reliable CID from the POTS line. This was for a SPA3102, not a 3000. I've never had a 3000, but everyone says they are nearly identical. I normally have 0 for both PSTN Answer Delay and PSTN Ring Thru Delay. Increasing the latter has also been said to solve this problem. However, if I change both of these values to 5 it does add a noticeable delay before any phones ring, but the CID remains unavailable. Perhaps this is because where I live (in the Netherlands) the local telco always sends the CID first. Thanks anyway! Cheers, Jaap ___ -- 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] SPA3102 registration problem
Quoting Jaap Winius [EMAIL PROTECTED]: My problem is that normal SPA3102 configurations just don't seem to work. I can't even get the FXS port to register. I'm beginning to suspect that my unit is defective. Today I called the vendor (voipsolutions.be) and was passed on to a knowledgeable tech support guy (!) who suggested that I configure a static IP address for the Internet gateway on the SPA3102 and use that instead of the LAN gateway. It worked! The registration problem is likely a bug, albeit an interesting one. Unfortunately, I'm still no better off using this device as a PSTN gateway than I am with the SPA3000, as I still can't get it to pass on the Caller ID. Cheers Jaap ___ -- 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] SPA3102 registration problem
Quoting Mandeep Singh Bhabha [EMAIL PROTECTED]: what i did to configure SPA3102 is ... My problem is that normal SPA3102 configurations just don't seem to work. I can't even get the FXS port to register. I'm beginning to suspect that my unit is defective. Here's why: If I configure the FXS port to register with my Asterisk server using the most basic sip.conf configuration *without* a password, then it does actually register. The only problem is the address it gives: bitis*CLI sip show peers Name/username HostDyn Nat ACL Port Status 8000/8000 127.0.0.1D 5060 OK (1 ms) That's right: instead of 192.168.1.8, it's telling Asterisk that it's available on the loopback address! I'll bet this is why it's not able to register using a password. I got the same results with three different firmware versions. Still, it's an unusual way for a device to be broken. Could there be another reason for this behavior? Otherwise I'll just have to try to explain it to the vendor and perhaps to Linksys tech support and ask for my money back. Cheers, Jaap ___ -- 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] SPA3102 registration problem
Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I did the SPA3000 and the Linksys PAP2T. For example, in all three cases this is the way I configured /etc/asterisk/sip.conf for Line 1: [4000] type=friend host=dynamic context=phones-m secret=1234 dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes The device is configured to register Line 1 with the SIP proxy and as a result the command sip show peers would eventually say the following: Name/username Host Dyn Nat ACL PortStatus 4000/4000 192.168.1.3 D 5060OK (13 ms) Not so with the SPA3102, in which case I always get: Name/username Host Dyn Nat ACL PortStatus 4000 (Unspecified) D 0 UNKNOWN After some tests, I found out that the SPA3102 is indeed trying to register, but that Asterisk seems to be ignoring it. Using tcpdump, I can see that registration packets are regularly being sent to the Asterisk server (bitis): 15:30:49.567288 IP spa3102.umrk.to.sip bitis.umrk.to.sip: SIP, length: 482 Eh% ...xREGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 12 15:30:49.568390 IP spa3102.umrk.to.sip-tls bitis.umrk.to.sip: SIP, length: 492 Eh%x... ...%REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 12 This sequence keeps on repeating. Also, if I change the sip.conf settings above to type=peer and host=192.168.1.3, I'll see these messages appear on the Asterisk console: [Feb 27 15:17:34] NOTICE[10893]: chan_sip.c:12414 handle_response_peerpoke: Peer '4000' is now Reachable. (7ms / 2000ms) [Feb 27 15:17:35] ERROR[10893]: chan_sip.c:8513 register_verify: Peer '4000' is trying to register, but not configured as host=dynamic [Feb 27 15:17:35] NOTICE[10893]: chan_sip.c:14943 handle_request_register: Registration from 'Margriet sip:[EMAIL PROTECTED]' failed for '192.168.1.3' - Peer is not supposed to register If, in this case, I configure the SPA3102 not to register any of its extensions, Asterisk will report them to be reachable and there won't be any more errors on the console, but in actual fact the extensions won't be available: I won't be able to call the phone attached to it due to congestion, and if I pick up that phone to make a call, I'll immediately hear a busy signal. What could be causing this situation? I'm using Asterisk 1.4.14 and the SPA3102 has the latest firmware version: 5.1.7(GW). I should also mention that I'm not interested in using this device's broadband router functionality. Any help would be much appreciated! Thanks, Jaap ___ -- 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] SPA3102 registration problem
Quoting Tim Johnson [EMAIL PROTECTED]: I see you put a password line in your sip.conf, but I do not see a username line. Also, you might want to check the port #'s for both the Line 1 and PSTN line. I use 5060 and 5061, respectively. Hopefully this either helps, or puts you on the right track. The username is 8000, so I don't believe it's necessary to mention it. As for the ports, I'm using them in the same way you suggest. Yet it refuses to work. My first attempt involved copying my SPA3000's working configuration to the SPA3102. That didn't work. So, I reset the device and applies a configuration generated by Voxilla's wizard, which worked for me with the SPA3000. Not that this has lead to any real differences, but it's still not working. There must be something else different about the SPA3102. I did see a problem with it mentioned somewhere in which it's connection with the local Asterisk server would fail (I think temporarily) when changes to the state of its Internet connection occurred (obviously not an issue with the SPA3000). I hope this has nothing to do with my problem. Thanks anyway! Cheers, Jaap ___ -- 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] Changing the automon output filename
Hi list,
The default automon (touch monitor) output file name format is:
auto-epoch-caller-callee.wav
A variable is available to modify the second half:
auto-epoch-${TOUCH_MONITOR}.wav
But, I can't modify the first half, 'auto-epoch-', with any variables
that I know of, including ${MONITOR_FILENAME}. I want to immediately
convert this output file to mp3, e.g. with ${MONITOR_EXEC_ARGS}, but I
can't because it's impossible to predict a file name that includes an
epoch number.
Can anyone help?
Thanks,
Jaap
___
-- 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] SPA-3000 caller ID and KPN
Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS lines and I've confirmed that is working. Yet, I've not been able to get the SPA-3000 to pas the CID on to Asterisk, and I know of no relevant SPA-3000 settings for doing this other than: * Regional Miscellaneous Caller ID Method: ETSI DTMF with PR * PSTN Line PSTN-To-VoIP Gateway Setup PSTN CID For VoIP CID: yes As some have suggested, I've also set the Regional / Miscellaneous / Caller ID FSK Standard: to 'bell 202', but this seems like nonsense to me as it should not make a difference once you've selected a DTMF CID method. I've also experimented with increasing the answer and ring-through delays, but this makes no difference. I've been told that this is because KPN always sends the CID on ahead of the rest of the call to begin with. Could it be that I'm missing something? Thanks, Jaap ___ -- 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] SPA-3000 caller ID and KPN
Quoting Tim Johnson [EMAIL PROTECTED]: I have a SPA3102 which is supposed to be similar. Make sure you leave the PSTN -- Subscriber Information -- Display Name blank. Also, in your sip.conf file, do not specify any callerid= value. ... It was worth a try, but unfortunately it makes no difference. Thanks anyway, though. Cheers, Jaap ___ -- 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] Fritz! Card/CAPI Help.
Quoting Razza [EMAIL PROTECTED]: I'm running - 2.6.23.15-137.fc8 which appears to be supported at ATrpms. I built a F8 box, added ATrpms to the repository list, executed yum -y install fcpci, which forced a kernel upgrade. Unfortunately, I can't test any of this since I'm running a Debian server. I installed the drivers/kernel drivers from ATrpms ( http://dl.atrpms.net/all/fcpci-03.11.07-14.fc8.i386.rpm and ... ). It seems you're in luck to some degree. I downloaded these files and examined the links within, but they simply point back to the AVM site that still offers the old driver versions. I guess your versions are a hack. Thanks for the heads up on the Cologne cards, if they work with just zaptel updates thats probably the easiest method, although Fritz! cards are available on ebay for about £5! The Cologne cards aren't expensive either: EUR 29,- to EUR 35,- in the Netherlands. I got one for only EUR 10,- recently. But even if the Cologne cards are more expensive than the Fritz! cards, and even if I could get the Fritz! cards to work on Debian again, it does indeed sound like the Cologne cards a lot easier to configure and maintain. Cheers, Jaap ___ -- 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] Fritz! Card/CAPI Help.
Quoting Axel Thimm [EMAIL PROTECTED]: There are patches inside that will work on Debian as well, just get the src.rpm and pick out the patches. Now, why am I not surprised? Actually, if I had known this back in early December, I'd be following your advice and thanking you now. I previously used these cards with Hylafax (and capi4hylafax), but the drivers supplied by AVM would not compile for kernel 2.6.18 after I upgraded to Debian etch. It was after I ran into this trouble that I was advised to use Cologne cards with Asterisk and IAXmodem instead; it was supposed to be a much more satisfying solution. I can't agree more! Okay, I still have to compile the Zaptel modules, but at least I no longer have to compile the fcpi module as well and pray that the next time a new kernel comes out I'll be able to repeat the process. Or, do you think the AVM Fritz!Card PCI has some advantage over an HFC-S Cologne card? Cheers, Jaap ___ -- 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] Fritz! Card/CAPI Help.
Quoting Razza [EMAIL PROTECTED]: Hi list, i'm keen to move to Asterisk 1.6, so really need to update my system which is running Mandrake 9.2 although it has been solid for years, fo Fedora 8. I have a Fritz! card for ISDN BRI, ... I 'modprobe capi' and 'modprobe fcpci' which appear to work fine, ... Interesting. I still have several AVM Fritz!Cards, but I stopped using them after I upgraded my server because I could no longer get them to work. I used to compile the fcpci module for kernel 2.6.8, but it doesn't work with 2.6.18. AFAIK, AVM haven't bothered to produce any new Linux code for these cards since July 2005 (even though they still sell them): ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/ Or, is newer fcpci code available from somewher else that will compile against later kernels? Your message seems to suggest this. Are you sure fcpci is loaded (lsmod)? I don't think the CAPI stuff will load without it. Anyway, I've since switched to using Cologne cards instead. You have to compile some Zaptel modules for them. These cards are cheaper, easier to obtain and work just fine. Cheers, Jaap ___ -- 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] Touch monitor file name format
Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
No, that doesn't work. ${MONITOR_FILENAME} can influence the filenames
in the string that you can tack onto the somix sequence using
${MONITOR_EXEC_ARGS}, but not the file name that automon produces. I
suppose you could also regard the automon output file name format as:
auto-${EPOCH}-${TOUCH_MONITOR}
The default is:
auto-${EPOCH}-caller-calee
Once again, it's easy to change and/or predict what the
${TOUCH_MONITOR} part is going to be, but AFAIK not the
'auto-${EPOCH}-' part. Therefore, if I'm right, there's no way to
manipulate the automon output using ${MONITOR_EXEC_ARGS}.
Thanks anyway,
Jaap
___
-- 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] MWI problem with Siemens Gigaset S675 IP
Quoting Steve Langstaff [EMAIL PROTECTED]: The 481 Call Leg/Transaction Does Not Exist response to the NOTIFY makes me think that you might need to configure the phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages from the phone when it is booted? Yeah, sure. And there are some error messages mixed in too: == 14:01:23.425955 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 473 ... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.426075 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 509 [EMAIL PROTECTED] ...vSIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:23.480238 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..k... ..F.SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.480375 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 14:01:23.918830 arp who-has gigaset.umrk.to tell bitis.umrk.to ../.E .. 14:01:23.921726 arp reply gigaset.umrk.to is-at 00:01:e3:77:f8:67 (oui Unknown) ...w.g../.E .. 14:01:24.539636 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 476 E.. ..2gSUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.539816 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 512 [EMAIL PROTECTED] ...ySIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:24.594442 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..i... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.594557 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 E...- [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 == Before this was a series of REGISTER messages, and afterwards a series of OPTIONS messages. However, no errors there. Also, this is without having set 'mailbox=1000' or '[EMAIL PROTECTED]' in /etc/asterisk/sip.conf. And, now that I look at it again, the network mailbox settings for the Siemens phone won't have anything to do with these errors either, since it simply makes it possible to associate a button on each handset with an extension used to access a voicemail account. Thanks, Jaap ___ -- 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] MWI problem with Siemens Gigaset S675 IP
Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. Thanks, Jaap ___ -- 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] MWI problem with Siemens Gigaset S675 IP
Quoting Henry Devito [EMAIL PROTECTED]: Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is) I've had to add the voicemail context to get MWI to work correctly in the past. According to the documentation, you shouldn't have to add @context if the context is 'default'. But, I went ahead and tried it out anyway. I even tried using some other context names, but it makes no difference: the error remains the same. Thanks anyway, Jaap ___ -- 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] Touch monitor file name format
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands
after the somix sequence for mp3 conversion. This should work, but
I've so far failed to produce any mp3 files because I'm not able to
predict the above epoch number. If I could alter 'auto-${EPOCH}-', or
if it was stored in a variable I could use, then I'm sure my plan will
succeed.
Any ideas?
Thanks,
Jaap
___
-- 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] Need good voicemail documentation
Quoting Jaap Winius [EMAIL PROTECTED]: After wrestling with the voicemail system for a while (Asterisk 1.4.14, Debian etch), I got it to work, but I still have lots of questions, like: * Why can't I delete any voicemail messages? (Response: Message undeleted.) * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response: I'm sorry, I did not understand your response.) * How come if I put [EMAIL PROTECTED] in my phone's context of sip.conf, do I get an error? (CLI: ...Remote host can't match request NOTIFY to call...) The first three problems were apparently due to a database connectivity problem. Earlier, I had set up an Asterisk database in postgresql and configured cdr_pgsql.conf, but neglected to do the same for res_pgsql.conf. Once that was taken care of, this voicemail weirdness disappeared. The fourth problem has to do with MWI and the SIP phone I'm using. Cheers, Jaap ___ -- 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] Automon reliability issue
Quoting Drew Gibson [EMAIL PROTECTED]: We made this function reliable by including the word quickly in our instructions for pressing the keycode to start the recording. ... Indeed, but somehow I don't think my users will be satisfied with that. Although a private confirmation beep to the initiator of the recording would be handy, this is the way things have to be in order to use the features.conf codes and still allow the use of * and # when calling outside IVR and voicemail systems. eg. Enter your password followed by the pound key... Understandable. A confirmation beep would therefore be an acceptable solution, not to mention a significant improvement for automon. Is there an easy way to achieve this with Asterisk v1.4.14, or will it perhaps appear in a future version? Thanks, Jaap ___ -- 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] Automon reliability issue
Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes host=dynamic [3000] ; Siemens Gigaset E455 wireless analog phone ; attached to Linksys PAP2T SIP adapter. type=friend secret=4321 context=phones-j dtmfmode=rfc2833 qualify=yes host=dynamic In /etc/asterisk/features.conf: [featuremap] automon = *1 In /etc/asterisk/extensions.conf: [phones-j] exten = 2000,1,Dial(SIP/2000,,wW) exten = 3000,1,Dial(SIP/3000,,wW) When I make or receive a call with either of these extensions, it's possible to start and stop recording by pressing *1. However, this only works if I press the two keys in quick succession; if I'm not fast enough, all I see is lots of the following console output: -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 In other words, unless I'm also monitoring the console, I can never be sure that Asterisk has actually started or stopped recording a call after I press these keys. Frequently, I first have to make several attempts. Is there some way to get automon to work reliably, or is the Monitor() function the only thing we can really count on to record calls? Thanks, Jaap ___ -- 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] Automon reliability issue
Quoting Doug Lytle [EMAIL PROTECTED]: featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation (default is 500 ms) courtesytone = local/stutter ; Sound file to play to the parked caller ; when someone dials a parked call ; or the Touch Monitor is ; Activated/Deactivated. Excellent! I set the latter to courtesytone = beep and now I've got something that's pretty close to ideal. The only way I think this could be improved is if there were different sounds for activation and deactivation. However, this is definitely good enough for now. Thanks, Doug! Cheers, Jaap ___ -- 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] Need good voicemail documentation
Hi list, After wrestling with the voicemail system for a while (Asterisk 1.4.14, Debian etch), I got it to work, but I still have lots of questions, like: * Why can't I delete any voicemail messages? (Response: Message undeleted.) * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response: I'm sorry, I did not understand your response.) * How come if I put [EMAIL PROTECTED] in my phone's context of sip.conf, do I get an error? (CLI: ...Remote host can't match request NOTIFY to call...) Unfortunately, none of the books and other documentation I've found on the subject goes into enough detail to provide answers to such questions. So, can anyone recommend some good Asterisk voicemail documentation that goes beyond merely scratching the surface? Cheers, Jaap ___ -- 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] Can't delete voicemail messages
Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option Press 7 to delete this message is available, but when I press 7 the response is always message undeleted and the message is still there. What could I be missing here? Thanks, Jaap ___ -- 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] Can't delete voicemail messages
Quoting Michiel van Baak [EMAIL PROTECTED]: On 00:38, Wed 06 Feb 08, Jaap Winius wrote: Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option Press 7 to delete this message is available, but when I press 7 the response is always message undeleted and the message is still there. What could I be missing here? Can you post the CLI logs from when that is happening ? All I see is a list of sound files appearing as they are played -- no error messages of any kind. However, the sound files that are listed immediately after I hit 7 are: -- SIP/1000-081fc028 Playing 'vm-deleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-undeleted' (language 'en') I only hear the second one. These are quickly followed by a list of the usual menu options (see the full CLI log below involving this same call). Cheers, Jaap =Begin CLI log== == Spawn extension (phones-j, 7000, 6) exited non-zero on 'SIP/1000-081fc028' -- Executing [EMAIL PROTECTED]:1] Answer(SIP/1000-081fc028, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/1000-081fc028, 1) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000-081fc028, [EMAIL PROTECTED]|s) in new stack -- SIP/1000-081fc028 Playing 'vm-youhave' (language 'en') -- SIP/1000-081fc028 Playing 'digits/5' (language 'en') -- SIP/1000-081fc028 Playing 'vm-Old' (language 'en') -- SIP/1000-081fc028 Playing 'vm-messages' (language 'en') -- SIP/1000-081fc028 Playing 'vm-onefor' (language 'en') -- SIP/1000-081fc028 Playing 'vm-Old' (language 'en') -- SIP/1000-081fc028 Playing 'vm-messages' (language 'en') -- SIP/1000-081fc028 Playing 'vm-opts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-first' (language 'en') == Parsing '/var/spool/asterisk/voicemail/default/1000/Old/msg.txt': Found -- SIP/1000-081fc028 Playing '/var/spool/asterisk/voicemail/default/1000/Old/msg' (language 'en') -- SIP/1000-081fc028 Playing 'vm-advopts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-repeat' (language 'en') -- SIP/1000-081fc028 Playing 'vm-next' (language 'en') -- SIP/1000-081fc028 Playing 'vm-delete' (language 'en') -- SIP/1000-081fc028 Playing 'vm-toforward' (language 'en') -- SIP/1000-081fc028 Playing 'vm-savemessage' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-deleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-undeleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-advopts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-repeat' (language 'en') -- SIP/1000-081fc028 Playing 'vm-next' (language 'en') -- SIP/1000-081fc028 Playing 'vm-delete' (language 'en') -- SIP/1000-081fc028 Playing 'vm-toforward' (language 'en') -- SIP/1000-081fc028 Playing 'vm-savemessage' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-goodbye' (language 'en') -- Executing [EMAIL PROTECTED]:4] Wait(SIP/1000-081fc028, 1) in new stack -- Executing [EMAIL PROTECTED]:5] Playback(SIP/1000-081fc028, vm-goodbye) in new stack -- SIP/1000-081fc028 Playing 'vm-goodbye' (language 'en') -- Executing [EMAIL PROTECTED]:6] Hangup(SIP/1000-081fc028, ) in new stack == Spawn extension (phones-j, 7000, 6) exited non-zero on 'SIP/1000-081fc028' =End CLI log ___ -- 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] Can't delete voicemail messages
Quoting Andy Doss [EMAIL PROTECTED]: File permission error? That is just my first guess. I am kind of new to Asterisk myself. The files are all in /var/spool/asterisk/voicemail/ where the asterisk user has read/write access to everything. Also, I see no error messages that would indicate a permission or access error. Thanks anyway, Jaap ___ -- 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] Monitoring calls on demand
Hi list, Recently I figured out how to automatically record (Monitor) both incoming and outgoing calls, which is handy. However, since this is not always desirable (or legal), can Asterisk be configured to start recording at some arbitrary point during a call, to be determined by the user, e.g. by entering a key combination, and then stopped later on in a similar fashion? Thanks, Jaap ___ -- 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] Asterisk 1.4 and ISDN-BRI support -- Solution
Quoting Jaap Winius [EMAIL PROTECTED]:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro,
kernel, modules, versions, config files).
Thanks to the support I received here I now have a working system, so
I thought I'd show my appreciation by posting my configuration here
for anyone who's interested.
Telco: KPN Telecom (Netherlands)
ISDN hardware: HFC-S PCI card (Cologne chip).
OS: Debian GNU/Linux stable (etch)
Kernel: 2.6.18-5-k7 (for an AMD Athlon CPU)
Relevant links in /etc/apt/sources.list:
deb http://updates.xorcom.com/rapid etch main
deb-src http://updates.xorcom.com/rapid etch main
Relevant installed debian packages:
asterisk 1.4.14~dfsg-0.4849
asterisk-config 1.4.14~dfsg-0.4849
asterisk-doc 1.4.14~dfsg-0.4849
asterisk-sounds-main 1.4.14~dfsg-0.4849
zaptel1.4.7.xpp.r5178-2
zaptel-firmware 1.4.7.xpp.r5178-2
zaptel-modules-2.6.18-5-k71.4.7.xpp.r5178-2+2.6.18.dfsg.1-17 *
zaptel-source 1.4.7.xpp.r5178-2
*) Compiled from zaptel-source using the command
m-a a-i zaptel.
Note: All of these packages are from xorcom.com. Debian etch
provides v1.2 of the Asterisk and Zaptel packages, which I
found to be too problematic.
Relevant loaded modules:
xpp89088 0
vzaphfc24984 3
zaptel185956 10 xpp,vzaphfc
firmware_class 10048 0
crc_ccitt 2560 1 zaptel
Note: The zaptel-modules package includes both the older zaphfc
and the newer vzaphfc modules. If genzaptelconf -d is run, both
get loaded, which is confusing at best. Therefore, I opted to
remove the older zaphfc module. I'm not sure the xpp and
firmware_class modules are necessary either: they also get loaded,
but don't seem to cause any trouble. Finally, I've found that the
modules I do need don't work properly unless they get loaded with
the genzaptelconf -d command. I guess that it loads them with
some parameters.
/etc/asterisk/zapata.conf:
[trunkgroups]
[channels]
context=isdn-in
language=en
overlapdial=yes
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callerid=asreceived
rxgain=4.5
txgain=-3
callgroup=1
pickupgroup=1
pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=0
internationalprefix=00
echocancel=yes
echotraining=100
echocancelwhenbridged=yes
faxdetect=incoming
immediate=no
group=1
switchtype=euroisdn
signalling=bri_cpe
channel=1-2
Note: I doubt all of these settings are absolutely necessary,
but this works for me.
Relevant parts of /etc/asterisk/extensions.conf:
[globals]
[general]
[isdn-in]
exten = isdn-in,1,Goto(0715134449,1)
exten = 0031715134449,1,Goto(0715134449,1)
exten = 0715134449,1,Dial(SIP/1000,30)
exten = 0715134449,n,Hangup()
[outgoing]
exten = _003171.,1,Dial(Zap/g1/${EXTEN},,r)
[internal]
exten = 1000,1,Verbose(1|Extension 1000)
exten = 1000,n,Dial(SIP/1000,30)
exten = 1000,n,Hangup()
[phones]
include = internal
include = outgoing
Note: In the dial command, Dial(Zap/g1/${EXTEN},,r), g1
corresponds to group=1 in /etc/asterisk/zapata.conf.
/etc/asterisk/indications.conf:
[general]
country=nl
[nl]
description = Netherlands
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/500,0/9500
dialrecall = 425/500,0/50
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425/500,0/50
Some diagnostic information:
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Note: These channels are (In use) because Asterisk is using them.
# cat /proc/interrupts
CPU0
0: 218203798IO-APIC-edge timer
6: 3IO-APIC-edge floppy
8: 1IO-APIC-edge rtc
9: 0 IO-APIC-level acpi
15:129IO-APIC-edge ide1
169: 95059844 IO-APIC-level skge
177: 10547626 IO-APIC-level libata
185: 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb2, ...
193: 3923488639 IO-APIC-level vzaphfc
201: 0 IO-APIC-level via82cxxx
NMI: 0
LOC: 218195472
ERR: 0
MIS
[asterisk-users] Channel fallback
Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a warning issued to the caller before the call is actually placed. Is this possible with Asterisk? If so, how? Thanks, Jaap ___ -- 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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What is the Dial command you use? Can you post the relevant part of your diaplan? exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) In addition: are you sure that there are channels set for group=0 ? Maybe try a channel directly: Zap/1 or Zap/2 instead of Zap/g0 . Aha! So, that's what /g*/ refers to: the group number specified in zatapa.conf! I set this to g0 after blindly following your advice of 27 December last year: So basically dial to Zap/g0/NUMBER and it should dial it to and it should dial it to your provider. The problem was that I had set group=1 in the zatapa.conf. Now that I've changed the dial command to Zap/g1/, the situation is vastly improved: == Primary D-Channel on span 1 up and -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f68b0, Zap/g1/[EMAIL PROTECTED]||r) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/[EMAIL PROTECTED] -- Channel 0/1, span 1 got hangup request, cause 28 -- Hungup 'Zap/1-1' This is when I try to make a call, which looks much better now. In conclusion, there was nothing wrong with the channels being described as in use in /proc/zaptel/*. However, the Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) error messages that I was getting as a result using the wrong group number in the command to dial out were misleading. If only Asterisk had been able to distinguish between congestion and a group that simply didn't exist. Of course, I still have a few other problems, but they're different and seem minor in comparison, so I won't mention them in this thread. This particular problem, however, is hereby solved. Many thanks, Tzafrir! Cheers, Jaap ___ -- 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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f68b0, Zap/g1/[EMAIL PROTECTED]||r) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/[EMAIL PROTECTED] Again, you're calling an incorrect number. You dial to a number that includes the string @channels. This is of course a number your telco does not know how to handle. Also you may need to add pridialplan = unknown in zapata.conf. At first I didn't understand what you meant by '@channels'? in your last response. Now I see. It was another example of me using someone else's config without knowing what it did. Now I can dial out via ISDN as well as receive calls, so there's no longer any need to start up any new threads either. At least, not immediately. :-) Thanks a million, Tzafrir! Cheers, Jaap ___ -- 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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: ... I get the wierd impression that either both modules somehow get interrupts from the two cards, or each module handles a different card. This hsouldn't happen. So try blacklisting one of them: I've already done something like that: removing the vzaphfc directory from /lib/modules/2.6.18-5-k7/misc/, running depmod, and then genzaptelconf -sdc nl. The rest of the modules loaded fine, but again all (3) channels were in use. Of course, I also tried putting vzaphfc back and removing zaphfc instead, but the results were the same. In both cases, I also tried running Asterisk with a minimal zapata.conf: switchtype=euroisdn signalling=bri_cpe_ptmp channel=1-2 Just for fun, I even tried this with signalling=bri_cpe_ptp, but then Asterisk starts without any Zaptel support. I'm running out of options here. It looks to me like the current versions of the software I'm using (Asterisk 1.4.14, Zaptel 1.4.7) just don't include working support HFC-S PCI cards. Yet, I've read that it has apparently worked in the past, so maybe I should try to downgrade to Asterisk 1.2.13 and Zaptel 1.2.11 (the versions that come with Debian stable). Thanks, Jaap ___ -- 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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What do you mean by In Use? # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) When Asterisk runs, they are (In use) - by Asterisk. I would hope so. I do see: # asterisk -rx 'pri show spans' PRI span 1/0: Provisioned, Up, Active and # asterisk -rx 'zap show channels' Chan Extension Context Language MOH Interpret pseudoisdn-in en default 1isdn-in en default 2isdn-in en default This is when I use the vzaphfc modules. However, any attempt to dial out via ISDN results in errors like: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f3220, Zap/g0/[EMAIL PROTECTED]||r) in new stack [Jan 11 00:12:15] WARNING[1354]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081f3220' status is 'CONGESTION' [Jan 11 00:12:15] NOTICE[1354]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081f3220' not posted In this case, the rule that I use in extensions.conf for dialing out is: exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) signalling=bri_cpe I've tried that, but unless I'm doing something else wrong (hopefully), using signalling=bri_cpe instead of signalling=bri_cpe_ptmp makes no difference. Thanks, Jaap ___ -- 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] Asterisk 1.4 and ISDN-BRI support
Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.7, also from xorcom.com), but with no luck: all three channels that are created when the zaphfc or vzaphfc module loads always change to an 'in use' state as soon as Asterisk starts up and so can't be used. I've received the exact same results on two different systems. Thanks, Jaap ___ -- 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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: (if you set in /etc/default/zaptel: ZAPBRI_SIGNALLING=bri you'll get that from genzaptelconf. If I create a file like this, I end up with signalling=bri_cpe instead of signalling=bri_cpe_ptmp. Anyway, either you use zaphfc or vzaphfc. The first one that loads takes everything. So far I have succeeded in starting up Asterisk without the zaphfc module (if channels 4-5 aren't defined in zapata.conf), but not without vzaphfc. Not having vzaphfc loaded always results in Asterisk starting up without Zaptel support. However, whether I run it with or without zaphfc, all of the available ISDN channels are always busy and the CLI still frequently shows Primary D-Channel on span 1 down messages. What do you see on /proc/interrupts ? CPU0 0: 25025934IO-APIC-edge timer 6: 3IO-APIC-edge floppy 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 15:129IO-APIC-edge ide1 169: 286747 IO-APIC-level skge 177:1014894 IO-APIC-level libata 185: 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb2, ... 193: 114700219 IO-APIC-level vzaphfc, zaphfc 201: 0 IO-APIC-level via82cxxx NMI: 0 LOC: 25024942 ERR: 0 MIS: 0 Which of those two modules you can't unload when Asterisk is running? If I declare all of the channels in zapata.conf, like this... channel = 1-2 channel = 4-5 then neither of the modules can be unloaded while Asterisk is running. If I comment out the first line then I can unload vzaphfc, while if I comment out the second I can unload zaphfc, so I guess this is how the channels are related to the modules. This makes sense, because when the zaptel modules are loaded with genzaptelconf -d (-d = hardware detection), vzaphfc is always loaded first. Regarding genzaptelconf -d, I've found that it is essential for me to run this command first before starting Asterisk. If not, Asterisk will start, but without Zaptel support. During system bootup, only the zaptel van vzaphfc modules are loaded by the kernel, which is not enough. Instead, genzaptelconf's hardware detection loads these modules in the following order: Module Size Used by xpp88512 0 zaphfc 12956 0 vzaphfc24312 0 firmware_class 9600 0 zaptel184740 3 xpp,zaphfc,vzaphfc This works. However, if I try to load these modules manually in the same order, Asterisk will start without Zaptel support. I don't know yet how genzaptelconf accomplishes this, but I suspect that it passes certain parameters to the zaptel and/or vzaphfc modules as it loads them. I say that because, after running genzaptelconf -d, it's possible to remove the xpp, zaphfc (if channels 4-5 are not declared) and firmware_class modules before starting up Asterisk and still have Zaptel support, although all of the Zaptel channels will still be busy. Furthermore, it is therefore not my impression that zaphfc is interfering with vzaphfc to cause all the zap channels to be busy. FYI, my current /etc/asterisk/zapata.conf is as follows: - [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 callgroup=1 pickupgroup=1 immediate=yes group=1 switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 channel = 4-5 - More information can be found in my previous posts in this thread. By the way, I've now duplicated my results on a new system with a different motherboard, a new HFC-S card and a fresh Debian etch install, etc., but unfortunately the results were exactly the same: all zap channels busy as soon as Asterisk starts. If anybody has a working Asterisk v1.4 configuration for ISDN-BRI using an HFC-S card and Zaptel software, I'd love to see it. Thanks, Jaap ___ --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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What is the output of: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active Do incoming calls work? Negative, and nothing shows up on the CLI. And that's after creating separate contexts called [default] and [pstn-in] in extensions.conf for incoming ISDN calls. Interesting... which one of those two is used? Good question. I've wanted to test that, but they're all the same: in use. I suspect vzaphfc is loaded automatically by udev, unless you have zaphfc explicitly in /etc/modules . It's not mentioned in /etc/modules. I also tried removing only vzaphfc or zaphfc and learned two more things: 1) After modifying zapata-channels.conf accordingly, no zap channels will show up in either of these configurations, i.e. pri show spans shows nothing. 2) If I start Asterisk by running genzaptelconf -sd -c nl, the other module will first get loaded, zapata-channels.conf will be restored to its original state* and all the channels will once again be in use. # cat zapata-channels.conf ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default ; Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default This is what genzaptelconf -sd -c nl keeps producing, although it doesn't look right. But, even if I comment out the first or second part and restart Asterisk, the remaining channels are always in use, dialing in doesn't work (number not available), and nor does dialing out (cause 34 - Circuit/channel congestion). Cheers, Jaap ___ --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] HFC-S zap channels always busy
Quoting Michiel van Baak [EMAIL PROTECTED]: I don't know about NL but in the UK, multiple ISDN2e lines have to be configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode? It's the same here in .nl Interesting, but I would think this to be unnecessary in my case, since I have only one ISDN-BRI line. It's just that for some reason the software keeps loading both the zaphfc and vzaphfc modules, which makes it look like I have two lines. But even if I do configure the system with signalling = bri_cpe_ptp, it makes no difference: all of the channels are still busy. Thanks anyway, though. Cheers, Jaap ___ --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] HFC-S zap channels always busy
Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f3698, Zap/g0/[EMAIL PROTECTED]||r) in new stack [Jan 3 15:32:06] WARNING[9769]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081f3698' status is 'CONGESTION' == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down Hopefully, someone here with more experience can point me in the direction of a solution. Here are hopefully some more clues: # lsmod | grep zap zaphfc 13660 1 vzaphfc24984 1 zaptel185956 9 xpp,zaphfc,vzaphfc crc_ccitt 2560 1 zaptel # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) It looks like the vzaphfc module creates a virtual interface. I have only one HFC-S PCI card installed. Each channel is (In use) immediately after Asterisk is started. CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefault en default 1from-pstn en default 2from-pstn en default 4from-pstn en default 5from-pstn en default CLI zap restart Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! Not a good idea, because that results in... CLI zap show channels Chan Extension Context Language MOH Interpret the channels disappearing altogether. However, I can restore the situation back to its original, albeit useless, state if I stop and start Asterisk. My configuration files are as follows: /etc/asterisk/zapata-channels.conf (after running genzaptelconf -sd -c nl): group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default /etc/asterisk/zapata.conf (supposed to work in the Netherlands): [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Abbreviated /etc/asterisk/extensions.conf: [globals] [general] [isdn-out] exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) [internal] exten = 1000,1,Verbose(1|Extension 1000) exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() [phones] include = internal include = isdn-out Any ideas? TIA, Jaap ___ --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] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What is the output of: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active Do incoming calls work? I haven't configured that yet. Interesting... which one of those two is used? Good question. I've wanted to test that, but they're all the same: in use. I suspect vzaphfc is loaded automatically by udev, unless you have zaphfc explicitly in /etc/modules . It's not mentioned in /etc/modules. Cheers, Jaap ___ --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] Zap channels for HFC-S PCI card not responding
Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? Thanks, Jaap PS -- Below is some info regarding my configuration. === Zaptel version: 1.4.7 (incl. firmware and modules). OS: Debian etch. Loaded modules: zaphfc 13660 1 vzaphfc24984 1 zaptel185956 9 xpp,zaphfc,vzaphfc crc_ccitt 2560 1 zaptel # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) # ztcfg -vv Zaptel Version: 1.4.7-Xorcom-trunk-r5178 Echo Canceller: MG2 Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) 6 channels to configure. /etc/asterisk/zapata-channels.conf after running genzaptelconf -sd -c nl: group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default /etc/asterisk/zapata.conf (supposed to work in the Netherlands): [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Abbreviated /etc/asterisk/extensions.conf: [globals] [general] [isdn-out] exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) [internal] exten = 1000,1,Verbose(1|Extension 1000) exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() [phones] include = internal include = isdn-out Any attempts to call out result in the following CLI output: [Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION' [Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081ff9f8' not posted CLI zap show channels: Chan Extension Context Language MOH Interpret pseudodefault en default 1from-pstn en default 2from-pstn en default 4from-pstn en default 5from-pstn en default CLI zap restart: Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! Not a good idea, since that results in... CLI zap show channels: Chan Extension Context Language MOH Interpret the channels disappearing altogether! === ___ --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] Zap channels for HFC-S PCI card not responding
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What do you mean by busy? What exactly do you see? This kind of thing: # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) Any attempts to call out result in the following CLI output: [Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION' [Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081ff9f8' not posted CLI zap restart: Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! This and more is from my previous message (sorry, that didn't just contain configuration information). Thanks, Jaap ___ --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] Problems with zaptel and HFC-S PCI card
Quoting Tzafrir Cohen [EMAIL PROTECTED]: ... this error that keeps appearing in my syslog and kern.log: zaphfc: empty HDLC frame or bad CRC received Try using the zaptel packages from: deb http://updates.xorcom.com/rapid etch main This upgraded Asterisk from v1.2 to v1.4.14 and the errors have disappeared. Thanks! Jaap ___ --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] Cirpack KeepAlive packets causing SIP errors
Hi list, After a recent upgrade to Asterisk v1.4.14, my message log is now filling up with the following error messages: - [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI --- SIP read from 82.101.62.99:5060 --- Cirpack KeepAlive Packet - Seeing as these packets are being sent by one of my service providers, I can't just turn them off. What's the best solution for this problem? Thanks, Jaap ___ --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] Cirpack KeepAlive packets causing SIP errors
Quoting Michiel van Baak [EMAIL PROTECTED]: - [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI --- SIP read from 82.101.62.99:5060 --- Cirpack KeepAlive Packet - Are you using XS4ALL ? Yes, although currently I'm mostly using InternetCalls for SIP. But I think these packets are indeed coming from a host (b3g6-nl.sip.b3g-telecom.com) associated with XS4ALL's VoIP service. http://svn.digium.com/view/asterisk?view=revrevision=93741 Maybe you can backport this change. Sounds like a good idea, but I'm having trouble getting the source code for Debian etch from xorcom.com to compile regardless. After attempting to compile with dpkg-buildpackage -rfakeroot -uc -b, it soon errors out with: Patch bristuff/ast-send-message does not remove cleanly \ (refresh it or enforce with -f) make: *** [unpatch] Error 1 Have I forgotten something? Thanks, Jaap ___ --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] Cirpack KeepAlive packets causing SIP errors
Quoting Michiel van Baak [EMAIL PROTECTED]: Sounds like a good idea, but I'm having trouble getting the source code for Debian etch from xorcom.com to compile regardless. I have no idea. I got it to compile. My mistake; I had attempted to modify chan_sip.c directly. It then refused to compile, but it also continued to after I had restored the original file, which threw me off. After a fresh download it compiled without any problems. I'll have to see if I can alter the code using quilt, as Tzafrir suggested, but otherwise I'll just follow Hans' advice and drop the incoming packets with this iptables rule: # drop Keep Alive packets from Cirpack SIP proxy xs4all /sbin/iptables -A INPUT -p udp -m udp --dport 5060 -m string --string Cirpack KeepAlive Packet --algo bm -j DROP (Thanks, Hans!) Cheers, Jaap ___ --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] Problems with zaptel and HFC-S PCI card
Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down Second, the syslog and the kern.log were quickly filling up with messages like these: Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, cpu throtteling enabled. Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. Dec 27 16:52:53 bitis last message repeated 31 times Dec 27 16:52:53 bitis kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=4069, z2=4062, wanted 8 got 7), probably a buffer overrun. Asterisk doesn't even have to be running for this to happen, but it can be brought to a halt by unloading the zaphfc module. I'm not aware of any CPU throttling on this system (an AMD Athon running at 1100 MHz). The OS is Debian etch running Linux kernel 2.6.18 (-5-k7). I've installed asterisk and asterisk-bristuff 1.2.13~dfsg-2etch2, as well as zaptel and zaptel-source 1.2.11.dfsg-1 to compile the necessary modules. My current configuration is as follows: cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) I think TE mode is fine, since I only need it to connect an outside line. Internally, I plan (hope) to use only SIP phones. /etc/asterisk/zapata.conf : [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=local prilocaldialplan=unknown nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Incidentally, this needs to work in the Netherlands. /etc/asterisk/zapata-channels.conf switchtype = euroisdn signalling = bri_net channel = 1-2 To connect to an outside line, I think signalling may need to be set to something else, but I'm not sure. The genzaptelconf shell script I used to produce it is buggy, so for all I know these settings may be wrong or even incomplete. /etc/asterisk/modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_capi.so load = res_musiconhold.so noload = chan_alsa.so [global] I've so far made no changes to extensions.conf to use the ISDN card. The linux modules zaptel, xpp and zaphfc get loaded automatically, but I haven't figured out yet from where. I'm thinking the zaphfc module may need to be loaded with a few (extra?) parameters before it starts behaving itself. Any help would be most welcome. Thanks! Jaap ___ --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] Problems with zaptel and HFC-S PCI card
Hi list, Just thought I'd let you know that the problems outlined in my previous post apparently had to do with a bad card. After swapping it out for another one the messages went away. Of course, I still have some problems. For instance, there's this error that keeps appearing in my syslog and kern.log: zaphfc: empty HDLC frame or bad CRC received Any idea how to get rid of it? Thanks, Jaap == Quoting Jaap Winius [EMAIL PROTECTED]: Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down Second, the syslog and the kern.log were quickly filling up with messages like these: Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, cpu throtteling enabled. Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. Dec 27 16:52:53 bitis last message repeated 31 times Dec 27 16:52:53 bitis kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=4069, z2=4062, wanted 8 got 7), probably a buffer overrun. ___ --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] ISDN BRI support with HFC-PCI cards?
Quoting Tzafrir Cohen [EMAIL PROTECTED]: cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F4) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 ACTIVATED (F7) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) Looks like it's already configured and used by Asterisk. Indeed. It would appear that Asterisk now recognizes these cards. Of course, I now have another set of problems, but I'll ask about those in a new thread. However, I believe that the ports are disconnected, right? Physically? One had a cable in it, the other one didn't. Cheers, Jaap ___ --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] ISDN BRI support with HFC-PCI cards?
Hi all, Could someone please point me in the direction of some reasonable instructions for setting up ISDN BRI support for Asterisk 1.2 (Debian etch) with HFC-PCI cards (Cologne chips) and/or cards with Winbond W6692CF chips? I keep finding solutions that involve running misdn-init. However, this approach seems to have been deprecated in favor of something else. Thanks, Jaap ___ --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] ISDN BRI support with HFC-PCI cards?
Quoting Tzafrir Cohen [EMAIL PROTECTED]: Could someone please point me in the direction of some reasonable instructions for setting up ISDN BRI support for Asterisk 1.2 (Debian etch) with HFC-PCI cards (Cologne chips) and/or cards with Winbond W6692CF chips? apt-get install asterisk zaptel-source m-a a-i zaptel echo #include zapata-channels.conf /etc/asterisk/zapata.conf genzaptelconf -sd That should basically do it. That looks promising, but am I right that this requires the zaptel package to be installed as well? Otherwise there's no genzaptelconf (or one that works). However, after installing the zaptel package, I get these errors: # genzaptelconf -sd Stopping Asterisk PBX: asterisk. cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory Starting Asterisk PBX: asterisk. # Looks like it's not creating its temporary directory in /tmp. Any ideas? Thanks, Jaap ___ --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] ISDN BRI support with HFC-PCI cards?
Quoting Tzafrir Cohen [EMAIL PROTECTED]: However, after installing the zaptel package, I get these errors: # genzaptelconf -sd Stopping Asterisk PBX: asterisk. cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory Starting Asterisk PBX: asterisk. cat /proc/zaptel/* Here's the output from that command: == Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F4) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 ACTIVATED (F7) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) == Thanks, Jaap ___ --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] sip.conf for internetcalls.com
Hi all, Perhaps someone here could help me with this. I'm new to Asterisk, but have already met with some success at getting my first system to work with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com. The config for the former works fine, but my InternetCalls.com config works only intermittently for incoming calls. It currently looks like this: [general] port=5060 bindaddr=0.0.0.0 srvlookup=yes register = usr-name:passwd@sip.internetcalls.com/inetcalls-in [inetcalls] type=friend context=inetcalls-in nat=yes username=usr-name fromuser=usr-name secret=passwd host=sip.internetcalls.com canreinvite=no qualify=yes dtmfmode=inband insecure=invite disallow=all allow=ulaw allow=alaw When I dial in from outside and it doesn't work, all I get is a busy signal and no further clues as to what might be wrong. Any ideas? Thanks very much, Jaap ___ --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.conf for internetcalls.com
Quoting Justin Case [EMAIL PROTECTED]: What comes up in the Asterisk CLI? When it's not working, nothing appears in the CLI even though I've used set verbose 10. Also it can be a NAT issue? How can that lead to this intermittent behavior? I've already set nat=yes. Also, I'm using an ADSL router with a NAT; not anything like iptables. Have Asterisk register every 3-4 minutes. I'm not sure how to do that. I found defaultexpirey, but the default for it is two minutes. Anyway, why would that help with Asterisk, when my previous SIP client, a Linksys SPA3000, was configured with a register expire time of an hour and worked fine with InternetCalls.com. I think something else is going on. Using tcpdump, I see this when things are working okay: -- 23:38:05.354523 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 847 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:38:05.355065 IP bitis.umrk.to.sip 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip: SIP, length: 471 FSIP/2.0 100 Trying Via: SIP/2.0/UDP 194.221.62.198:50 . 23:38:11.007350 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 507 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- The ACK packet is sent after the conversation (.) has ended. However, when it doesn't work, I see this: -- 23:42:24.736377 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 841 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:42:24.736898 IP bitis.umrk.to.sip 194.120.0.198.sip: SIP, length: 445 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 194.120.0.198: 23:42:24.756967 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 505 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- In this case, the ACK follows immediately after the 404 Not Found. Cheers, Jaap ___ --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
