[asterisk-users] When line in use busy signal?!
hello all, how isit possible to give a busy signal if the line is in use? For me it is ringing and signaling on my phone, when i call out and i get another call. My hint is this: exten = 31,hint,SIP/1000131SIP/1000131a i have one softphone an one hardphone Regards René -- René Enskat Internet-Administrator Teamware GmbH Stahlgruberring 11 D-81829 München Tel: 089-427005.31 Fax: 089-427005.55 E-Mail: [EMAIL PROTECTED] http://www.tmwr.de/ http://www.tmwr.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + door opener
Jerry wrote: Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. I'd be interested too, I was thinking of upgrading our door opener with a telephone line adapter and an FXO port from the linecard, but if I can do this without using an FXO port (and doesn't cost the earth) It would be great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Now
Ciao Carlos, I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. I have had some problems with Asterisk Now, until I switched to text mode installation, and then everything ran smoothly. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about sip account format
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango: How about: exten = _X.,1,Answer Does it include all numbers and characters? As of the docs, no. It should only match 0123456789 See http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnected after 1 hour
Klaverstyn, David C wrote: There seems to be something in Asterisk that disconnects the call at 1 hour. At 59 minutes there is a beep then 1 minute later the call is dropped. I have a basic install Asterisk Ver. 1.2.13. I have not specifically said that calls are to be disconnected at a certain time (not that I know how to do that). Well, it could be that you have the L() option in your Dial string. Or also, if your I connected to a PBX via E1 or something, they usually cut calls at 1 hour. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Am Mittwoch, den 20.12.2006, 14:42 -0500 schrieb Doug Crompton: Anthony, Ok I understand. The 011 is unique though and I guess the problem is the length of the remaining digits. This could vary based on country?? and I suspect there is no unique rule that could be applied??? I have not studied this but is there any uniqness to the remaining digits? Doug There are no general rules for international number lengths. In certain countries, the numbering plan is very specific about how long a telephone number is - the US is the best example, where ANY phone number is area(3)+line(7). AFAIK Luxembourg and a few countries with a small number of telephones have rules as well. On the contrary, in Germany there are area codes between 2 digits (only a few, Hamburg, Berlin, Munich, Frankfurt) and 5 digits, and inside those cities numbering varies wildly. Old lines (registered pre-1960 or so) sometimes still have 3-digit numbers, especially in the countryside where there is no urge to assign new phone numbers. A friend of mine has the numbers 328 and 1653990 on the same ISDN line. And then, there are DIDs with varying number length. A company I worked for years ago had 9559-X where X might be 0 for central, two-digit 1X for department calling groups, [234]XX for individual phones and 9XXX for individual fax numbers. No rules there, bad luck. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Match a Numer - then continue with, dialplan
RL == Richard Lyman [EMAIL PROTECTED] writes: RL grr, i hate when i typo (and reply to my own posts) exten = RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) Heh, if you want to chase typos, perhaps you should add an underscore before ? /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101 with Unicall
Carlos Chavez wrote: CAS signalling on span 1 conflicts with HDLC with FCS check on channel My guess is not to use HDLC, as the error says above, that it conflicts with CAS. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clear ast database
Rilawich Ango wrote: Any command to refresh or clear the whole ast database? asterisk -rx 'stop now' rm astdb asterisk -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101 with Unicall
On Thu, 21 Dec 2006 08:33:27 -0500, Doug Lytle wrote Carlos Chavez wrote: CAS signalling on span 1 conflicts with HDLC with FCS check on channel My guess is not to use HDLC, as the error says above, that it conflicts with CAS. I wish it were that easy and obvious. I only found one setting on the configuration to disable HDLC and it is disabled. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Insert 1+areacode for VOIP calls
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls.
I'd like to get around this if possible. The following is what I have
in my extensions.conf..
exten = _1NXXNXX,1,Set(CALLERID(num)=6162997590)
exten = _1NXXNXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten =
_1NXXNXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500)
exten =
_1NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
Is there a way I can create a _NXX extension and insert 1 and
areacode when dialing?
Any help appreciated,
Phil
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Re: [asterisk-users] Insert 1+areacode for VOIP calls
Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?
exten = _NXX,1,Set(CALLERID(num)=6162997590)
exten = _NXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1514${EXTEN})
replace 514 with your area code
hth
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Re: [asterisk-users] Insert 1+areacode for VOIP calls
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match NXX-NXX- and pop a one in place of what you dialed.
Alex
On 12/21/06, Phil Finkler [EMAIL PROTECTED] wrote:
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls. I'd
like to get around this if possible. The following is what I have in my
extensions.conf..
exten = _1NXXNXX,1,Set(CALLERID(num)=6162997590)
exten = _1NXXNXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten = _1NXXNXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500)
exten = _1NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?
Any help appreciated,
Phil
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--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] Insert 1+areacode for VOIP calls
Phil,
Yeah, I just realized that I didn't answer your question. Time Bandit did
though, look at his solution!
Alex
On 12/21/06, Alex Robar [EMAIL PROTECTED] wrote:
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match NXX-NXX- and pop a one in place of what you dialed.
Alex
On 12/21/06, Phil Finkler [EMAIL PROTECTED] wrote:
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls. I'd
like to get around this if possible. The following is what I have in my
extensions.conf..
exten = _1NXXNXX,1,Set(CALLERID(num)=6162997590)
exten = _1NXXNXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten =
_1NXXNXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500)
exten =
_1NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
Is there a way I can create a _NXX extension and insert 1 and
areacode when dialing?
Any help appreciated,
Phil
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--
Alex Robar
[EMAIL PROTECTED]
--
Alex Robar
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[asterisk-users] AELPARSE - Wish/Suggestion
I was playing with aelparse last night and I thought it would be nice if the output of the it's operation was a little more structured. I've written a app that allows me to edit ael/conf files from a windows environment and upload them to the asterisk box, commit a reload, restart, etc, etc. This is performed through a socket server running on the asterisk box. One of the things I was wanting to do was to be able to transfer an ael file to the asterisk box, run aelparse (running aelparse -q -d) and pull the output from stdout back over to the windows client for inspection. As it stands, I can parse through the output, especially looking for ael_yyparse strings, etc for errors. But I'm assuming that the output could/would changes over time. If the output is not likely to change that much, it's no problem to parse the output as it is, but having maybe a standard dump file (like using the -w option) produced would be nice and wouldn't need to change very much over time. XML file would be great, but I'd be game for anything that is structured a bit more and not likely to change. Any thoughts? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AELPARSE - Wish/Suggestion
On Thu, Dec 21, 2006 at 09:57:50AM -0500, Lee Jenkins wrote: I was playing with aelparse last night and I thought it would be nice if the output of the it's operation was a little more structured. I've written a app that allows me to edit ael/conf files from a windows environment and upload them to the asterisk box, commit a reload, restart, etc, etc. This is performed through a socket server running on the asterisk box. One of the things I was wanting to do was to be able to transfer an ael file to the asterisk box, run aelparse (running aelparse -q -d) and pull the output from stdout back over to the windows client for inspection. As it stands, I can parse through the output, especially looking for ael_yyparse strings, etc for errors. But I'm assuming that the output could/would changes over time. If the output is not likely to change that much, it's no problem to parse the output as it is, but having maybe a standard dump file (like using the -w option) produced would be nice and wouldn't need to change very much over time. XML file would be great, but I'd be game for anything that is structured a bit more and not likely to change. Any thoughts? Maybe an optional different file descriptor rather than a dump file? Would that have been of more use to you? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Insert 1+areacode for VOIP calls
; Dial wether long distance is preceeded by 1 or not
; Dial LD via gizmo
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _1NXXNXX,2,Macro(failann,${DIALSTATUS})
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _NXXNXX,2,Macro(failann,${DIALSTATUS})
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RE: [asterisk-users] Need quality toll free 800 number over IAX?
One way audio is almost always caused by firewalls / NAT translation. Since there is neither on IPKall, my guess would be to look at the other end. With 20k + users, most have succeeded in correcting this problem via their hardware / software. I encourage you to look at the user forum for some suggestions. IPKall http://voxilla.com/PNphpBB2-viewforum-f-38.html IPKall Forum _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: Wednesday, December 20, 2006 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need quality toll free 800 number over IAX? I have used www.ipkall.com I have had one way audio for two weeks now with no reply from CS. So I will back you up on this I guess http://www.kall8.com/ would be the same I think they are one in the same. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBX http://www.bochterservices.com/?j=PBXt=email t=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdid http://www.bochterservices.com/?t=TFdidt=email t=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMS http://www.bochterservices.com/?t=VMSt=email t=email --For new and used security items http://www.bochterservices.com/?j=store http://www.bochterservices.com/?j=storet=email t=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=gold http://www.bochterservices.com/?j=goldt=email t=email Kevin Walsh wrote: www.IPKall.com mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Thomson ST2030S and BLF
I too am wondering if someone has a contact at Thomson, some of the softkeys need to either be fixed or have the option to remove (like FwdVM and Pickup keys). In addition, has anyone notice a humming noise when using the handset? I can hear it and so can the person that I am calling. On 12/21/06, Alberto Pastore [EMAIL PROTECTED] wrote: Olivier ha scritto: ... I didn't get any usable reply yet, beside usual maybe with next release. From http://bugs.digium.com/view.php?id=5014, I don't think one key call pickup is going to appear anytime soon with Asterisk. Hi Olivier. That's a pity. ST2030s is in my opinion one of the best SIP phones, with all features a phone needs (very good provisionig support, poe, double ethernet, line keys, subscriber keys, remote phonebook, audio quality...) compared to its low price. It would be so easy to issue an INVITE to the very same key extension and do the pickup via Pickup() dialplan function... Do you have a direct contact with Thomson guys? I've tried to reach them on e-mail or phone but with no success... Anyway, thanks again for the NOTIFY call-id patch tip. That's a new toy to play with for a couple of day before giving up. Alberto. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AELPARSE - Wish/Suggestion
Tzafrir Cohen wrote: Maybe an optional different file descriptor rather than a dump file? Would that have been of more use to you? That could certainly work. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan
Benny Amorsen wrote: RL == Richard Lyman [EMAIL PROTECTED] writes: RL grr, i hate when i typo (and reply to my own posts) exten = RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) Heh, if you want to chase typos, perhaps you should add an underscore before ? /Benny if that were the case, then you should also change the CALLERID(name/num) references to macros! eh merry christmas to all! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Now
I think its rPath Linux, based on redhat. I've had some problems with Asterisk Now. My X100P card was not recognized since it didnt show in the zap channels in the GUI thats why I switched back to debian and install Asterisk from source. On 12/20/06, Carlos Alperin [EMAIL PROTECTED] wrote: I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] more than 32 callgroups pickupgroups
callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] more than 32 callgroups pickupgroups
I'm no C programmer, but is this 32 limit just an array definition somewhere? Wouldn't it be a no brainer to track it down and increase it so some very large number? -Original Message- From: John Harragin [mailto:[EMAIL PROTECTED] Sent: Thursday, December 21, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] more than 32 callgroups pickupgroups callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crashed
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
res_musiconhold.c:180
#5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c:1382
#6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405
#7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857
#8 0x08069293 in ast_generic_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c)
at channel.c:3524
#10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0,
config=0xb6c4feb0) at res_features.c:1319
#11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0,
peerflags=0xb6c50568) at app_dial.c:1577
#12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at
app_dial.c:1619
#13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 op05_x, exten=0xb659ff14 00116,
priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553
#14 0x0808efea in __ast_pbx_run (c=0xb659fcd0) at pbx.c:2227
#15 0x0808fcdf in pbx_thread (data=0xb7ed1900) at pbx.c:2514
#16 0xb7f7cb63 in start_thread () from /lib/tls/libpthread.so.0
#17 0xb7e7718a in clone () from /lib/tls/libc.so.6
another one:
#0 0xb6ff38e2 in decodeMP3 () from /usr/lib/asterisk/modules/format_mp3.so
#1 0xb6ff4be6 in key () from /usr/lib/asterisk/modules/format_mp3.so
#2 0xb6ff4545 in key () from /usr/lib/asterisk/modules/format_mp3.so
#3 0x0806d3a1 in ast_readframe (s=0xb7eb490c) at file.c:570
#4 0xb7b0c134 in moh_files_generator (chan=0xb6b26dc0, data=0xb6b03328, len=0,
samples=160) at res_musiconhold.c:246
#5 0x08068cfe in generator_force (data=0xb6b26dc0) at channel.c:1401
#6 0x08061c50 in ast_read (chan=0xb6b26dc0) at channel.c:1857
#7 0x08069293 in ast_generic_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#8 0x080655fd in ast_channel_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c)
at channel.c:3524
#9 0xb78ddd29 in ast_bridge_call (chan=0xb6b26dc0, peer=0x8699fe8,
config=0xb6677eb0) at res_features.c:1319
#10 0xb7033301 in dial_exec_full (chan=0xb6b26dc0, data=0xb6677eb0,
peerflags=0xb6678568) at app_dial.c:1577
#11 0xb7031dc5 in dial_exec (chan=0x48, data=0x48) at app_dial.c:1619
#12 0x0808e445 in pbx_extension_helper (c=0xb6b26dc0, con=0x48, context=0xb6b26f10 op05_x, exten=0xb6b27004 00116, priority=1,
label=0x0, callerid=0x0, action=0) at pbx.c:553
#13 0x0808efea in __ast_pbx_run (c=0xb6b26dc0) at pbx.c:2227
#14 0x0808fcdf in pbx_thread (data=0x48) at pbx.c:2514
#15 0xb7f5fb63 in start_thread () from /lib/tls/libpthread.so.0
#16 0xb7e5a18a in clone () from /lib/tls/libc.so.6
here's the versions of various components:
asterisk: 1.2.7.1, zaptel: 1.2.5, libpri: 1.2.2
any clues would be appreciated?
--
Edwin Lam [EMAIL PROTECTED]
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Ph: +1 415 439 4988 Fax: +1 415 283 3370
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Yes thats the bottom line, its mostly the country code which can be 1-3 digits long. There is no rules based solution for this. Historicaly each country picked a number out of a hat except the US (which had to be number 1) because as we all know it's the centre of the universe. The former USSR had to go for 7 and Russia still kept this after it's break-up. All the other former USSR countries have settled on a 3 digit number but (as far a I know) can still be accessed by dialing 7. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Am Mittwoch, den 20.12.2006, 14:42 -0500 schrieb Doug Crompton: Anthony, Ok I understand. The 011 is unique though and I guess the problem is the length of the remaining digits. This could vary based on country?? and I suspect there is no unique rule that could be applied??? I have not studied this but is there any uniqness to the remaining digits? Doug There are no general rules for international number lengths. In certain countries, the numbering plan is very specific about how long a telephone number is - the US is the best example, where ANY phone number is area(3)+line(7). AFAIK Luxembourg and a few countries with a small number of telephones have rules as well. On the contrary, in Germany there are area codes between 2 digits (only a few, Hamburg, Berlin, Munich, Frankfurt) and 5 digits, and inside those cities numbering varies wildly. Old lines (registered pre-1960 or so) sometimes still have 3-digit numbers, especially in the countryside where there is no urge to assign new phone numbers. A friend of mine has the numbers 328 and 1653990 on the same ISDN line. And then, there are DIDs with varying number length. A company I worked for years ago had 9559-X where X might be 0 for central, two-digit 1X for department calling groups, [234]XX for individual phones and 9XXX for individual fax numbers. No rules there, bad luck. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message- From: Richard Lyman [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan Douglas Garstang wrote: -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] *snipped David, this is completely different from what I am trying to do. Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? Doug. if it is just callerid then wouldn't the gf stuff (if it still exists) work? it was something like (man i'm getting old, looking up in wiki) exten = s,1,Answer() exten = s,,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) exten = s,2,Set(CALLERID(name)=INSIDE NAME|CALLERID(num)=xx) exten = s,3,Dial(yadda) would obviously be the callerid num of the internal exten I don't think that scales to hundreds of companies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP-2000 and Asterisk Configuration
Hello, We are having a hard time making the GXP-2000 work reliably with Asterisk. We have several clients using the GXP-2000. These phones are behing NAT and our Asterisk server has a public IP (no NAT). The biggest problem we face is the clients complain of random, but frequent, calls (in or out) where there is no audio. We have enabled STUN on the phones and when we do a sip show peer X, we see that the Addr-IP and Reg. Contact fields both have the public IP of each respective client. When we do a sip debug peer X, we notice that Asterisk displays Warning: 399 207.XXX.XXX.XXX detected NAT type is symmetric NAT. We also noticed that some of our other clients show a different debug information Warning: 399 72.XXX.XXX.XXX detected NAT type is port restricted cone. Even worse, for some for example, one client that has multiple GXP-2000, some of them report the NAT as being symmetric and some report it as being port restricted. The phones are Grandstream GXP2000 1.1.1.14 and we are running Asterisk 1.2.13. What should be the proper setting in Asterisk for the nat option in the sip peer? What should be the proper settings in the GXP-2000 phone itself? I was recommended on the list that I use nat=route and to disable the STUN server on the GXP-2000. However, even though that seemed to work at first, it still presented the problem of calls with no audio. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + door opener
FYI, astribanks all come with outputs that can be used for door openers, combined with this product from Vikingelectronics.com that plugs into any fxs port you should have a complete solution for a door: http://www.vikingelectronics.com/products/view_product.php?pid=99 They (viking) has a door opener that plugs into FXO as well (C2000-A) On 12/21/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Jerry wrote: Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. I'd be interested too, I was thinking of upgrading our door opener with a telephone line adapter and an FXO port from the linecard, but if I can do this without using an FXO port (and doesn't cost the earth) It would be great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] more than 32 callgroups pickupgroups
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote: I'm no C programmer, but is this 32 limit just an array definition somewhere? Wouldn't it be a no brainer to track it down and increase it so some very large number? I think pickupgroup is defined as 'unsigned int' somewhere in channels.h. 32 is the number of bits in a 4-byte integer, so it's probably using a bitmask to define which pickupgroups a channel belongs to. I suppose if you are on a 64bit machine/os you /could/ try to make it a 64 bit pointer, but you should really check the source a bit more to see how exactly it's accessed (I didn't!) I don't know any .32bit integers on 32bit machines. -Original Message- From: John Harragin [mailto:[EMAIL PROTECTED] Sent: Thursday, December 21, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] more than 32 callgroups pickupgroups callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT: Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan
Douglas Garstang wrote: -Original Message- From: Richard Lyman [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan Douglas Garstang wrote: -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] *snipped David, this is completely different from what I am trying to do. Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? Doug. if it is just callerid then wouldn't the gf stuff (if it still exists) work? it was something like (man i'm getting old, looking up in wiki) exten = s,1,Answer() exten = s,,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) exten = s,2,Set(CALLERID(name)=INSIDE NAME|CALLERID(num)=xx) exten = s,3,Dial(yadda) would obviously be the callerid num of the internal exten I don't think that scales to hundreds of companies. hey doug, did you forget you already replied to this with If there's hundreds of companies on this box, we'd need an exponentially larger number of statements... not to mention it was YOU that defined it as Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? note: 2 companies tagged OT: as this doesn't even have a direction anymore. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXW-4108 8 port FXO
Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inform callers on recorded/monitored number.
Eric Jacksch wrote: You might also want to look at what the legal situation is in your jurisdiction. Here one only needs the consent of one party to the call, so I don’t have to advise the callee that the call is recorded if the caller consents to the recording. If you are in the U.S., you may find the following of interest: http://www.rcfp.org/taping/ -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXW-4108 8 port FXO
I would be very interested in getting an 8 port FXO myself. They are very new so I don't think there are any used ones out there yet. Does anybody out there in Canada stock them yet? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX calls not ringing
Greetings folks. I seem to be having a problem where calls made from an IAX device (three single-line phones attached to IAXys) do not play the ring tone when calling out. There's a dial tone when I pick up the phone, and the call goes through just fine, it just doesn't ring. All my SIP phones ring normally, however. Is there an option I need to enable that I'm missing? Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Thomson ST2030S and BLF
Andrew Joakimsen ha scritto: I too am wondering if someone has a contact at Thomson, some of the softkeys need to either be fixed or have the option to remove (like FwdVM and Pickup keys). In addition, has anyone notice a humming noise when using the handset? I can hear it and so can the person that I am calling. Honestly, I'm experiencing a good audio quality, no humming noise or hiss. Well, I'm using g711a... Alberto. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXW-4108 8 port FXO
Chris, These devices are still very new to the market. Finding reviews on them may be tough still. From our experience its a good little device for the dollar; but, keep in mind, it's still a low cost gateway and that normally means don't expect too much. We've sold few cases here and response on them have been both good and bad. Grandstream for a while stopped shipping any units they had due to firmware problems on the unit. Quality of this product is driven a lot off of the firmware alone unfortunately as well as the environment the units are being used in. However, supposedly Grandstream began shipping the unit again recently and released new firmware on many of their products which they have boasted to us as being much better. That being said, I can't say whether this unit works great, good, or poorly at this time. I would say it could work very well for a budget conscious,small office or home office setup. I'd like to see some good user reviews on this unit from people that have it running in a live environment. We've only played with it here. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Dec 21, 2006, at 3:29 PM, cb wrote: Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] International dialplans for Asterisk?
Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the 011 is the maximum number, because 17 is just not found: OK:1234567890123456 http://www.google.com/search?q=011X Not OK:12345678901234567 http://www.google.com/search?q=011XX This fellow includes 011x.T in his dial plan. Will this provide for more than 16 numbers?http://www.vovida.org/pipermail/mgcp/2003-November/001848.html Does anyone have some definitive sources for this subject? Other info: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg37207.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
On 21/12/06, Doug [EMAIL PROTECTED] wrote: Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the 011 is the maximum number, because 17 is just not found: OK:1234567890123456 http://www.google.com/search?q=011X Not OK:12345678901234567 http://www.google.com/search?q=011XX Why would you imagine that people in non-US countries would list their phone numbers on their websites in US International dialing format? Especially when more countries use '00' for their outbound international prefix than use '011'. As has already been mentioned recently, at least one country (Germany) has no hard limit on the length of a number - extra digits after the base number are delivered to the CPE for internal routing - kind-of self-administered DDI ranges. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX calls not ringing
On 16:03, Thu 21 Dec 06, Jay Moore wrote:
Greetings folks.
I seem to be having a problem where calls made from an IAX device (three
single-line phones attached to IAXys) do not play the ring tone when
calling out. There's a dial tone when I pick up the phone, and the call
goes through just fine, it just doesn't ring. All my SIP phones ring
normally, however. Is there an option I need to enable that I'm missing?
try the r option in the dial statement:
exten = s,10,Dial(IAX2/${DEVICE_TO_RING},45,tr)
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] Help with silence or gating of speech?
Hi, I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E for analog trunks ( extensions) plus some Polycom 501 601 phones. I have a problem in that the audio via the Polycoms is gated or muted during quiet parts of the other person's speech. This results in the start of words being clipped and quiet sounds being lost altogether. It would be OK if this was at a really low level, but it seems to take a fair but of audio before anything gets through - even quieter parts of 'on-hold' music from the trunk side are lost. I've spent ages searching using numerous combinations of words and cannot find any reference to this problem, or even figure out just what part of the overall system is causing it.. Any help appreciated! Robert Jenkins. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
On 22:56, Thu 21 Dec 06, Peter Bowyer wrote: On 21/12/06, Doug [EMAIL PROTECTED] wrote: Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the 011 is the maximum number, because 17 is just not found: OK:1234567890123456 http://www.google.com/search?q=011X Not OK:12345678901234567 http://www.google.com/search?q=011XX Why would you imagine that people in non-US countries would list their phone numbers on their websites in US International dialing format? Especially when more countries use '00' for their outbound international prefix than use '011'. As has already been mentioned recently, at least one country (Germany) has no hard limit on the length of a number - extra digits after the base number are delivered to the CPE for internal routing - kind-of self-administered DDI ranges. As far as I can remember (and our ITSP is telling us to do) the 'dial international' code will be gone soon. In our case we have to provide the number like this: country coderegionendpoint[extra digits] So for a dutch number you send: 31318787243 31 == .nl 318 == my local region 787243 == my endpoint I see this more and more. not only ITSP, also PSTN providers and cellphone providers. cellphone providers use this most of the time: +countryregionendpont The above number looks like: +31318787243 Try to get that from your telco, it makes life way more easy. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clear ast database
you mean we need to remove astdb manual? Totally restart asterisk even the whole server doesn't do the removement? On 12/21/06, Doug Lytle [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Any command to refresh or clear the whole ast database? asterisk -rx 'stop now' rm astdb asterisk -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about sip account format
Thanks. I got it. On 12/21/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango: How about: exten = _X.,1,Answer Does it include all numbers and characters? As of the docs, no. It should only match 0123456789 See http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with SUSE 10.2 and Sangoma A104D
Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in /dev/zap(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : ) This mine motherboard, has a NIC Gbps and if I leave qualified, the A104D not recognized for motherboar, if I disable the NIC Gbps, through lspci, I identify the A104D. But report this error in compiling the wanpipe (./Setup install) WANPIPE DRIVER COMPILE LOG Fri Dec 22 01:19:40 BRST 2006 --- make -C /lib/modules/2.6.18.2-34-default/build SUBDIRS=/usr/src/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make -C ../../../linux-2.6.18.2-34 O=../linux-2.6.18.2-34-obj/i386/default modules CC [M] /usr/src/wanpipe/kdrvtmp/sdladrv_src.o CC [M] /usr/src/wanpipe/kdrvtmp/sdladrv_fe.o /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:106: warning: 'sdla_te1_write_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:141: warning: 'sdla_te1_read_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:219: warning: 'sdla_shark_te1_write_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:292: warning: 'sdla_shark_te1_read_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:471: warning: 'sdla_shark_analog_write_fe' defined but not used LD [M] /usr/src/wanpipe/kdrvtmp/sdladrv.o Building modules, stage 2. MODPOST CC /usr/src/wanpipe/kdrvtmp/sdladrv.mod.o LD [M] /usr/src/wanpipe/kdrvtmp/sdladrv.ko make[1]: Leaving directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make -C /lib/modules/2.6.18.2-34-default/build SUBDIRS=/usr/src/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make -C ../../../linux-2.6.18.2-34 O=../linux-2.6.18.2-34-obj/i386/default modules CC [M] /usr/src/wanpipe/kdrvtmp/wanmain.o /usr/src/wanpipe/kdrvtmp/wanmain.c: In function 'wanrouter_ioctl': /usr/src/wanpipe/kdrvtmp/wanmain.c:546: error: 'struct inode' has no member named 'i_private' make[4]: *** [/usr/src/wanpipe/kdrvtmp/wanmain.o] Error 1 make[3]: *** [_module_/usr/src/wanpipe/kdrvtmp] Error 2 make[2]: *** [modules] Error 2 make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make: *** [all] Error 2 CLI ztcfg -v Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected My hardware is: Processor Pentium IV 3.0 1M 800 775 Motherboard PENTIUM IV ASUS P5VD2 -MX 775 1GB RAM Kingston DDR HD 80.0 GB IDE W.DIGITAL 7200RPM Please, help me! Best Regards Josue ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
I second that. I'm quite happy with the IPKall.com did number I use today. Only once in the last year was it unavailable when I needed it. So, not bulletproof, but good enough for me to use all day when I work at home. On 12/21/06, www.IPKall.com [EMAIL PROTECTED] wrote: One way audio is almost always caused by firewalls / NAT translation. Since there is neither on IPKall, my guess would be to look at the other end. With 20k + users, most have succeeded in correcting this problem via their hardware / software. I encourage you to look at the user forum for some suggestions. IPKall IPKall Forum http://voxilla.com/PNphpBB2-viewforum-f-38.html -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Al Bochter *Sent:* Wednesday, December 20, 2006 4:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Need quality toll free 800 number over IAX? I have used www.ipkall.com I have had one way audio for two weeks now with no reply from CS. So I will back you up on this I guess http://www.kall8.com/ would be the same I think they are one in the same. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Kevin Walsh wrote: www.IPKall.com [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Help with SUSE 10.2 and Sangoma A104D
Hi All. Forgive me, but mine motherboard is ASUS P5GPL-X SE Thank's Best Regards Josue 2006/12/22, Josué Conti [EMAIL PROTECTED]: Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5, sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in /dev/zap(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : ) This mine motherboard, has a NIC Gbps and if I leave qualified, the A104D not recognized for motherboar, if I disable the NIC Gbps, through lspci, I identify the A104D. But report this error in compiling the wanpipe (./Setup install) WANPIPE DRIVER COMPILE LOG Fri Dec 22 01:19:40 BRST 2006 --- make -C /lib/modules/2.6.18.2-34-default/build SUBDIRS=/usr/src/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make -C ../../../linux-2.6.18.2-34 O=../linux-2.6.18.2-34-obj/i386/default modules CC [M] /usr/src/wanpipe/kdrvtmp/sdladrv_src.o CC [M] /usr/src/wanpipe/kdrvtmp/sdladrv_fe.o /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:106: warning: 'sdla_te1_write_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:141: warning: 'sdla_te1_read_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:219: warning: 'sdla_shark_te1_write_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:292: warning: 'sdla_shark_te1_read_fe' defined but not used /usr/src/wanpipe/kdrvtmp/sdladrv_fe.c:471: warning: 'sdla_shark_analog_write_fe' defined but not used LD [M] /usr/src/wanpipe/kdrvtmp/sdladrv.o Building modules, stage 2. MODPOST CC /usr/src/wanpipe/kdrvtmp/sdladrv.mod.o LD [M] /usr/src/wanpipe/kdrvtmp/sdladrv.ko make[1]: Leaving directory `/usr/src/linux- 2.6.18.2-34-obj/i386/default' make -C /lib/modules/2.6.18.2-34-default/build SUBDIRS=/usr/src/wanpipe/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make -C ../../../linux-2.6.18.2-34 O=../linux-2.6.18.2-34-obj/i386/default modules CC [M] /usr/src/wanpipe/kdrvtmp/wanmain.o /usr/src/wanpipe/kdrvtmp/wanmain.c: In function 'wanrouter_ioctl': /usr/src/wanpipe/kdrvtmp/wanmain.c:546: error: 'struct inode' has no member named 'i_private' make[4]: *** [/usr/src/wanpipe/kdrvtmp/wanmain.o] Error 1 make[3]: *** [_module_/usr/src/wanpipe/kdrvtmp] Error 2 make[2]: *** [modules] Error 2 make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/linux- 2.6.18.2-34-obj/i386/default' make: *** [all] Error 2 CLI ztcfg -v Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected My hardware is: Processor Pentium IV 3.0 1M 800 775 Motherboard PENTIUM IV ASUS P5VD2 -MX 775 1GB RAM Kingston DDR HD 80.0 GB IDE W.DIGITAL 7200RPM Please, help me! Best Regards Josue ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect many fax lines?
We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these fax ports. The big question is what's the best way to connect these fax ports to *? 1) We could use an 8-port T1 card and link 6 Rhino FXS channel banks. 2) We could put 6 MultiTech MultiVOIP FX-24 boxes on the * LAN. 3) or ??? Anyone do something like this before? Any suggestions? I personally like the simplcity of the MultiVOIP boxes, plus the fact that they don't require T1 ports. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect many fax lines?
stay away from foip stick with channel banks On 12/21/06, Allen Casteran [EMAIL PROTECTED] wrote: We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these fax ports. The big question is what's the best way to connect these fax ports to *? 1) We could use an 8-port T1 card and link 6 Rhino FXS channel banks. 2) We could put 6 MultiTech MultiVOIP FX-24 boxes on the * LAN. 3) or ??? Anyone do something like this before? Any suggestions? I personally like the simplcity of the MultiVOIP boxes, plus the fact that they don't require T1 ports. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. Doug On Fri, 22 Dec 2006, Michiel van Baak wrote: The above number looks like: +31318787243 Try to get that from your telco, it makes life way more easy. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] asterisk + door opener
Hello the list, You can use FXS and em signalling to reverse the line polarity temporary to trigger an external door opener interface. This is very easy. Good Luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Thomas Kenyon Envoyé : jeudi 21 décembre 2006 12:13e. À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] asterisk + door opener Jerry wrote: Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. I'd be interested too, I was thinking of upgrading our door opener with a telephone line adapter and an FXO port from the linecard, but if I can do this without using an FXO port (and doesn't cost the earth) It would be great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about astdb
I noticed that asterisk will keep the phone record in astdb when the phone (especially hardphone) unplugged. After unplug the phone, I still get the phone information in astdb: database showkey SIP/Registry/1234 /SIP/Registry/1234 : 10.14.43.31:40876:60:1234:sip:[EMAIL PROTECTED]:40876;rinstance=09d900e954f2e92d I wonder whether asterisk will have a mechanism to check and remove the account without register message after period of time? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
I think the + convention started off because different countries have different international access codes. Well, on GSM networks, + can be a part of the number to represent the international access code ( the traditional access code in India is 00 for international). So to call Digium, from my GSM phone, I can use 0018775468963 or +18775468963 and Allison will answer :) Rajeev On 12/22/06, Doug Crompton [EMAIL PROTECTED] wrote: Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. Doug On Fri, 22 Dec 2006, Michiel van Baak wrote: The above number looks like: +31318787243 Try to get that from your telco, it makes life way more easy. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
