Re: [Asterisk-Users] Swedish CallerID?
On Sun, 3 Jul 2005, Josef Seger wrote: I have one other Dect phone connected to Digiums Card(TDM400P), an Ericsson DT 260. The Ericsson phone only supports true swedish standard CallerID (DTMF signalling before the first ring), and CallerID does not work for this phone:( I have measured the outgoing signal from the TDM400P card and I have confirmed thet NO DTMS signals is sent out. Is it possible to show swedish callerid on ordinary analog phones connected to the Digium card? If yes, can somebody see the problem in my configuration files? See the bug at http://bugs.digium.com/view.php?id=3866. The original poster did not respond in time. Perhaps you can help debug the patch there? If so, ask one of the maintainers to reopen the bug report. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repost: how to configure asterisk user and group rights
I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. 2. what directories and files it must have read and right access to 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Are these some info sources which go into these areas in depth? Obelix This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] #include not working with *1.0.9
We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions-phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context If we reload and restart *, and then try to dial one of the defined extensions in the included directory, nothing...just Service Unavailable. If I copy and paste a few of the extensions that are in the .conf files directly into the [default] context of the extensions.conf file, the extensions work. So it seems to me that the include statement no longer works in 1.0.9 I figure this is the case, because we were running 1.0.5 and the same config file worked fine. Anybody know what's going on? Brent This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include not working with *1.0.9
Just a thought, but I seem to recall that in the dialplan, inlcude and other similar statements are not prefixed by the hash character (#). Try include = . -Bryce On Jul 4, 2005, at 00:05, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions- phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context If we reload and restart *, and then try to dial one of the defined extensions in the included directory, nothing...just Service Unavailable. If I copy and paste a few of the extensions that are in the .conf files directly into the [default] context of the extensions.conf file, the extensions work. So it seems to me that the include statement no longer works in 1.0.9 I figure this is the case, because we were running 1.0.5 and the same config file worked fine. Anybody know what's going on? Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Repost: how to configure asterisk user and group rights
On Mon, Jul 04, 2005 at 06:55:42AM +, 8 hours after posting exactly the same message, Obelix wrote: I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. man asterisk. There is a switch -U 2. what directories and files it must have read and right access to Hmmm Try running it as a different user. When you get permissions denied it probably means you forgot something. 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Did you bother searching the wiki? http://voip-info.org/tiki-index.php?page=Asterisk+non-root -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include not working with *1.0.9
On Jul 4, 2005, at 12:05 AM, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions-phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context That happened to me in Jan or Feb of this year; just happened to be that on one particular day, the source I CVSed out had a broken * shell expander. I waited a day or two, redownloaded and recompiled and all was well. -- Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include not working with *1.0.9
On Jul 4, 2005, at 12:17 AM, Bryce Chidester wrote: Just a thought, but I seem to recall that in the dialplan, inlcude and other similar statements are not prefixed by the hash character (#). Try include = . -Bryce You're thinking of contextual includes, not filesystem includes -- which do use the hash. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include not working with *1.0.9
On Mon, Jul 04, 2005 at 02:05:17AM -0500, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions-phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context The support for globbing in #include has been merged into HEAD, and is also part of the Rapid packages. Either grab the debs from http://tzafrir.org.il/rapid108/ or use the debs source: deb http://tzafrir.org.il/rapid108 unstable/ or extract debian/patches//80_rapid-globinclude.dpatch from the asterisk diff in the above directory and apply it to the asterisk source you build. Despite my repetetive requests, the Debian package maintainers have not yet included this patch :-( -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Did the Broadvoice patch break asterisk ?
I think it might have. My understanding is BV does not require username and secret for incoming calls from BV to Asterisk, so a patch was written and then included in Asterisk to fix this. I have been testing for 4 weeks now, and have been shot down once in the digium bug tracker and pretty much blown off, but I'll try again here to see if I get anywhere. I and others noticed that we get pretty weird and annoying interactions between incoming Broadvoice calls and Asterisk. Specifically... The CDR is wrong, shows the wrong Broadvoice Number (the username in the trunk setup) The Asterisk system follows the wrong context for the trunk. After a LOT of testing what I seemed to have found is that Asterisk has been programmed to cache the information for the first incoming Broadvoice call, the number and the context. All subsequent calls from Broadvoice will use this number and context REGARDLESS of what the other Broadvoice trunks have been programmed with no matter how wrong the settings are for them, no errors, just chugs along using the info for the first trunk that rings in. So if the first trunk has a context that tells it to use the auto attendant, then ALL BV trunks will use the AA even if their context is to do something else. You can give all the other BV trunks a non-existent context and their is no error when they ring in, they just follow the valid context from the first trunk that rang in. The work aorund has been to use the /EXT option in the registration, but that is not a good fix, as other interactions are occuring as well. I have 14 BV trunks, and I have repeatedly proven that Asterisk will ignore ALL cutomizations for their username and context and instead will follow the settings for the first one that rings in following a reboot. Am I missing something here, or is this indeed a problem ? Our logs files and CDR are all screwed up and meaningless with respect to tracking back to the actual SIP CHANNEL activity as they always reflect the first trunk that rang in for all BV trunk activity. I am using the most current release of Asterisk. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Epia C3 Linux
Hello AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a normal hdd. Can that cause any problems? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Epia C3 Linux Michel Brabants wrote: Tzafrir Cohen wrote: On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote: Anyone know a good distro for an Epia Mobo with the C3 chip? Debian, as for any hardware :-p Heya, I have heard that epia C3 has full i586-support, but i686 support is not complete. greetings, Michel Wiley (and others), If you intend to run Asterisk (I hope so because this is asterisk-users :) ), AstLinux runs like a dream on the CS EPIA boards: http://www.astlinux.org AstLinux runs so well because everything was compiled for i586-MMX and higher processors. The same disclaimer applies to all of my other AstLinux posts: I am the creator/maintainer so I am obviously biased! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #include not working with *1.0.9
Tzafrir, Do you have patch description file which explains what the different patches do? Thanks, Brent Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Jul 04, 2005 at 02:05:17AM -0500, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions-phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context The support for globbing in #include has been merged into HEAD, and is also part of the Rapid packages. Either grab the debs from http://tzafrir.org.il/rapid108/ or use the debs source: deb http://tzafrir.org.il/rapid108 unstable/ or extract debian/patches//80_rapid-globinclude.dpatch from the asterisk diff in the above directory and apply it to the asterisk source you build. Despite my repetetive requests, the Debian package maintainers have not yet included this patch :-( -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
does this ring a bell for anyone?
On 7/3/05, no name [EMAIL PROTECTED] wrote:
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA - zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
so then I instructed asterisk to treat this as zap interface:
box:/etc/asterisk# egrep -v '(^;|^$)' zapata.conf
[channels]
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
immediate=no
priindication=outofband
overlapdial=no
usecallerid=yes
rxgain=0.0
txgain=0.0
context=inbound
callerid=asreceived
group=1
channel=1-2
defined this in the dialplan:
office:/etc/asterisk# tail -21 extensions.conf| egrep -v '(^;|^$)'
[inbound]
; my main number is 1234567,
; I am using 3-digit internal extensions
exten = _11234567XXX,1,Goto(internal-phones,$EXTEN:8,1)
exten = _XXX,1,Goto(internal-phones,${EXTEN},1)
; this acts as catch-all so dialling just the main number goes to x200
exten = s,1,Answer
exten = s,2,Goto(internal-phones,200,1)
now when I call e.g. 1234567200 from the outside, asterisk sees this as:
-- Extension '20' in context 'inbound' from '1some other number'
does not exist. Rejecting call on channel 0/1, span 1
why does asterisk INSIST on chopping the trailing digit off the
dialled number? I don't get it.
please help!
Bernie
--
best,
Bernie
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Re: [Asterisk-Users] #include not working with *1.0.9
On Mon, Jul 04, 2005 at 03:13:26AM -0500, [EMAIL PROTECTED] wrote: Tzafrir, Do you have patch description file which explains what the different patches do? Extrat the asterisk_*.diff.gz using zcat that_file.diff.gz | patch -p1 in an empty directory. This will create a subdirectory debian with all the debian files in it. debian/patches/ should have all the actual source patches. Those will be dpatch executables, which are actual uniform diffs that the standard patch can use, but with a small header. In that header there should be a short description (DP:) although not all dpatches include one. I tryed to remember to add one to ones I have added. Also search for the name of the patch file in the debian/changelog file. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Sound (2nd post)
Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error messages. My problem Is that none of the sounds work, there is no sound for any of the following features 1. Voicemail prompts 2. the menu macro in Dial 3. Music on hold 4. conversation Here's everything I have tried so far. 1. update fedora (I have compiled asterisk off the disk release and also after Redhat updates) 2. update Asterisk ( I have recompiled several times over the past month with different HEAD versions) 3. recompile mpg-123 using both 'r' and 'q' versions I am getting a console message from time to time which say Application asterisk uses obsolete OSS audio interface But Dial, Voicemail and Music on hold report no errors. The production system works fine on the older CVS head from Jan 26 2005. With an out of date Fedora install off the CDs. Thanks Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Idefisk iax2 softphone - new version
We just released a new version of the idefisk iax2 softphone, version 1.21 beta, available for download at http://www.asteriskguru.com/tools/idefisk_beta.php Some bugs were fixed, some new bugs might have been introduced :) - The problem with delays is finally gone!!! (one of the bugs was a memory leak, everybody using an older version is encouraged to upgrade.) Privacy Warning: Version 1.21 of the softphone will send 'usage statistics' to the asteriskguru webserver, this can be disabled in the configuration menu (uncheck send usage statistics). The only info sent is the version of idefisk used. Many thanks to digium, stevek and others for the iaxlib and iaxclient libraries. Zoa. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT : Wengo sucks
Would just like to warn everybody for Wengo.fr Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird ring back
Hi Yair, Thanks for your email. Unfortunately no reply or response from anyone yet. Please let me know if you hear anything - I'm also battling to resolve the problem. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Yair Hakak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 03, 2005 10:26 AM Subject: Fwd: [Asterisk-Users] Weird ring back Hi David, I am having the same problem and i am wondering if you have been able to solve this. any help you can give me is appreciated. thanks, yair -- Forwarded message -- From: David Wilson [EMAIL PROTECTED] Date: Jun 22, 2005 10:15 PM Subject: [Asterisk-Users] Weird ring back To: asterisk asterisk-users@lists.digium.com Hi guys, I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. Anyone had this before ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MAKEing zaptel and ztdummy on SuSE 9.3 - Repost
Hi, Sorry to re-post, but I'm still having hassles with ztdummy. I'm using kernel 2.6.11.4-21.7-smp and Asterisk 1.0.8 on SuSE 9.3 The first 3 makes (see below) for zaptel work out ok - but the ztdummy.ko (etc) files *are* created even though I haven't yet uncommented ztdummy in Makefile. If the ztdummy bits are compiled, why edit Makefile run make again? modprobe ztdummy fails. I'm following the instructions on Asterisk Doc Proj: astersik_1.0.8/docs/docs-html_one/vm1.html#AEN30 and they don't seem to work out. The order suggested is: cd zaptel make clean make linux26 make install vi Makefile (and uncomment ztdummy) make modprobe zaptel modprobe ztdummy Here are the relevent problems: gl0:/usr/src/zaptel-1.0.8 # lsmod | grep usb usbserial 34024 0 usbcore 121688 4 usbserial,ehci_hcd,uhci_hcd gl0:/usr/src/zaptel-1.0.8 # lsmod | grep z Module Size Used by gl0:/usr/src/zaptel-1.0.8 # modprobe zaptel gl0:/usr/src/zaptel-1.0.8 # modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.11.4-21.7-smp/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy gl0:/usr/src/zaptel-1.0.8 # lsmod | grep z Module Size Used by zaptel185860 0 crc_ccitt 6144 1 zaptel gl0:/usr/src/zaptel-1.0.8 # the end bit of dmesg returns: zaptel: unsupported module, tainting kernel. Zapata Telephony Interface Registered on major 196 ztdummy: unsupported module, tainting kernel. ztdummy: disagrees about version of symbol zt_receive ztdummy: Unknown symbol zt_receive, st_info == 0x1 ztdummy: disagrees about version of symbol zt_transmit ztdummy: Unknown symbol zt_transmit, st_info == 0x1 ztdummy: disagrees about version of symbol zt_unregister ztdummy: Unknown symbol zt_unregister, st_info == 0x1 ztdummy: disagrees about version of symbol zt_register ztdummy: Unknown symbol zt_register, st_info == 0x1 load_module: err 0xfffe (dont worry) I'd be grateful for some pointers. TIA, Zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 12, Issue 17
Hello, they are successful to start asterisk, task that the error that I had previously had had to a configuration problem. Start asterisk in modality consol and when two softphone speaks is not felt well, and I have the following error: -- Registered SIP '1000' at 10.0.0.7 port 5060 expires 1800 -- Saved useragent X-Lite release 1103m for peer 1000 -- Registered SIP '1001' at 10.0.0.5 port 5060 expires 1800 -- Saved useragent X-Lite release 1103m for peer 1001 -- Executing Dial(SIP/1001-73df, sip/1000|20|rt) in new stack -- Called 1000 -- SIP/1000-60e3 is ringing -- SIP/1000-60e3 answered SIP/1001-73df -- Attempting native bridge of SIP/1001-73df and SIP/1000-60e3 Jul 4 10:20:14 NOTICE[840]: rtp.c:281 process_rfc3389: Comfort noise support inc omplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP : 10.0.0.5 as I can make __ TISCALI ADSL 1.25 MEGA a soli 19.95 euro/mese Solo con Tiscali Adsl navighi senza limiti di tempo a meno di 20 euro al mese. E in piu' telefoni senza pagare il canone Telecom! Scopri come, clicca qui http://abbonati.tiscali.it/adsl/sa/1e25flat_tc/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT Mark Spencer lunch in Paris Fri July 8th
There is going to be another great Paris lunch with Mark this Friday. The restaurant will probably be in the southern part of Paris in the 14th arrdt. like last time. Please contact me off list if you are able to attend. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls
Bernie Ott wrote:
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
Try an extension with four digits and one with two. You may see, that *
chops all after 10 digits!!!
bye
Ronald
does this ring a bell for anyone?
On 7/3/05, no name [EMAIL PROTECTED] wrote:
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA - zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
so then I instructed asterisk to treat this as zap interface:
box:/etc/asterisk# egrep -v '(^;|^$)' zapata.conf
[channels]
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
immediate=no
priindication=outofband
overlapdial=no
usecallerid=yes
rxgain=0.0
txgain=0.0
context=inbound
callerid=asreceived
group=1
channel=1-2
defined this in the dialplan:
office:/etc/asterisk# tail -21 extensions.conf| egrep -v '(^;|^$)'
[inbound]
; my main number is 1234567,
; I am using 3-digit internal extensions
exten = _11234567XXX,1,Goto(internal-phones,$EXTEN:8,1)
exten = _XXX,1,Goto(internal-phones,${EXTEN},1)
; this acts as catch-all so dialling just the main number goes to x200
exten = s,1,Answer
exten = s,2,Goto(internal-phones,200,1)
now when I call e.g. 1234567200 from the outside, asterisk sees this as:
-- Extension '20' in context 'inbound' from '1some other number'
does not exist. Rejecting call on channel 0/1, span 1
why does asterisk INSIST on chopping the trailing digit off the
dialled number? I don't get it.
please help!
Bernie
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[Asterisk-Users] Fax DETECTION with CAPI
hi, I have dabian sarge with asterisk 1.0.7 and chan_capi 0.3.5 with AVM fritz card. I would like use detecion of fax, but it don't work. So, i would like know if it's possible to work fax detection with this card? And if it's possible how?? Thanks you for your help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco 7920
Hi; Is it possible for me to use my cisco7920 with Asterisk in any way? Cheers Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7920
Sure! http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2 regards, roland From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoglu Sent: Monday, July 04, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] cisco 7920 Hi; Is it possible for me to use my cisco7920 with Asterisk in any way? Cheers Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : Wengo sucks
Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'. That's been true of every cellphone and Internet company I've used in France: FT, Noos, Cégétel, Orange... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls
Hi Ronald,
* chopping after 10 digits is fine - our number is 12345673 digit
ext though so there's a total of 9 digits.
On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Bernie Ott wrote:
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
Try an extension with four digits and one with two. You may see, that *
chops all after 10 digits!!!
bye
Ronald
does this ring a bell for anyone?
On 7/3/05, no name [EMAIL PROTECTED] wrote:
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA - zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)
HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
so then I instructed asterisk to treat this as zap interface:
box:/etc/asterisk# egrep -v '(^;|^$)' zapata.conf
[channels]
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
immediate=no
priindication=outofband
overlapdial=no
usecallerid=yes
rxgain=0.0
txgain=0.0
context=inbound
callerid=asreceived
group=1
channel=1-2
defined this in the dialplan:
office:/etc/asterisk# tail -21 extensions.conf| egrep -v '(^;|^$)'
[inbound]
; my main number is 1234567,
; I am using 3-digit internal extensions
exten = _11234567XXX,1,Goto(internal-phones,$EXTEN:8,1)
exten = _XXX,1,Goto(internal-phones,${EXTEN},1)
; this acts as catch-all so dialling just the main number goes to x200
exten = s,1,Answer
exten = s,2,Goto(internal-phones,200,1)
now when I call e.g. 1234567200 from the outside, asterisk sees this as:
-- Extension '20' in context 'inbound' from '1some other number'
does not exist. Rejecting call on channel 0/1, span 1
why does asterisk INSIST on chopping the trailing digit off the
dialled number? I don't get it.
please help!
Bernie
--
best,
Bernie
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[Asterisk-Users] Dial *97 to pickup voicemail buts says my password incorrect
Hello I am at extension 200 and I know there is a voicemail message waiting. I dial *97 and am prompted for the password. I enter 1234 which I have set as my voicemail password. What can I do to troubleshoot? Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7920
Thanks for link roland. The document mentions about firmware for Cisco 7920 ? Is it SIP firmware ? I asked the cisco seller for Sip firmware but they said sip firmware is unavailable for 7920? Would Creating files mentioned on the document be enough for configuration? Thanks again -Original Message- From: Roland Zagler [mailto:[EMAIL PROTECTED] Sent: Monday, July 04, 2005 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] cisco 7920 Sure! http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2 regards, roland From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoglu Sent: Monday, July 04, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] cisco 7920 Hi; Is it possible for me to use my cisco7920 with Asterisk in any way? Cheers Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial *97 to pickup voicemail buts says my passwordincorrect
I have found that if I dial from another extension *98 and select extn 200 and enter password 1234 it works. So is it something to do with configuration on my IP Phone? It is a Grandstream GXP2000 running: Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3 Anyone got any ideas? Angus - Original Message - From: Angus Comber To: asterisk-users@lists.digium.com Sent: Monday, July 04, 2005 12:20 PM Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my passwordincorrect Hello I am at extension 200 and I know there is a voicemail message waiting. I dial *97 and am prompted for the password. I enter 1234 which I have set as my voicemail password. What can I do to troubleshoot? Angus Comber Itel Office Software Ltd 5 Enmore Gardens London, SW14 8RF Tel: 020 8878 7367 Fax: 020 8876 7257 Em: [EMAIL PROTECTED] web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp fon behind NAT gw
Hi I've a mgcp fon (swissvoice IP10s) behind a NAT router. Configured is NAT for both in/out going on port 2427. Now I got the following mgcp debug messages when i try mgcp audit endpoint endpoint -- from 172.16.98.57:2427 Verb: 'RSIP', Identifier: '5346', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Retransmitting: 200 5346 OK to 192.168.2.3:2427 MGCP read: RSIP 5346 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart - IMHO means 200 everything ok. But what means the RM: restart ? For me it look's like the asterisk knows the NAT gw and also the EP. But the EP can't find the Call Agent. It's clear at the display. waiting for call agent... the mgcp.conf looks like this -- [192.168.2.3] ;router als RGW context=default host=192.168.2.3 nat=yes line = aaln/1 callerid=1423 port 2427 is bound. Directly connected via switch there's no problem. May be anyone have some hints or tipps... thx in advance regards mathias roehl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On Mon, Jul 04, 2005 at 05:19:39PM +0800, Ronald_Wiplinger wrote: Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! How do you know it is not connected? Why do you assume it should be connected? Please answer your questions, and while you do: verify all of your assumptions. After you've answered them, please try to guess what our next question would have been. Is there a sip peer or it in sip.conf? How does that sip peer appear in 'sip show peers' on the CLI? voip-info,org, google, and such are valueble resources for answering the questions. For example: connected basically means (for a SIP client) being registered as a SIP peer. Though a client can technically connect without registrating in advance. So: is there a section for it in sip.conf? How does it appear in the output of 'sip show peers'? Do you have any reason to believe that the grandstream phone is actually sending any packets to your asterisk computer? Try running: tcpdump -n 'host IP_ADDRESS_THE_PHONE' on your asterisk system. Ethereal may be useful for protocol analisys, but tcpdump is great if you just want to know if there is traffic. Naturally another thing to try is to eliminate one part of the problem: can you use a different SIP client with the same definitions of the server (or vice-versa)? Does that SIP client work with any other SIP server? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Robert Goodyear wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! Do you have qualify=1000 or some value in the sip.conf? Are you getting a time when you do a sip show peers? It could be the phone is registering and then losing network, and if the registration time is an hour it would still show as registered even if it was uncontactable. (I think). IANAAE (I am not an asterisk expert.) e.g. 212/ 192.168.0.25 D 255.255.255.255 5062 OK (24 ms) 211/ 192.168.0.25 D 255.255.255.255 5060 OK (27 ms) 210/ 192.168.0.23 D 255.255.255.255 5060 OK (59 ms) 203/ (Unspecified)D 255.255.255.255 0UNKNOWN Regards Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Intel Blade Machine...
Hello, I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? I would appreciate for giving me feedback regarding this issue. Regards Nahid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wi-fi phone advice
The one that looks identical is selling at $180 from www.broad-tel.com/index_en.php On 7/1/05, Richard Malcolm-Smith [EMAIL PROTECTED] wrote: If it does materialize, im up for 3 or 4 of them at that price. Huddleston, Robert wrote: Well poo - if I can use that word I'm one of those poor family guys who loves to buy hardware on the cheap =) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Connection to Splicecom Maximiser
Don't know about your asterisk box, but i've made Max talk to an IPOffice (Avaya) using H323. in brief, it was a case of going to modules in the Max and adding a 'virtual trunk module' , and in the IPOffice create an IP Trunk, most of the fields in both systems are self evident - especially if you know anything about linking IPOffice. Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Intel Blade Machine...
Nahid Hossain wrote: I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? We've had Asterisk running on a blade for some time. Blades as such can be used but with a couple of restrictions: - There's probably no room for PCI cards, so no zap hardware - Check the kind of USB supported on the board (UHCI vs OHCI, for ztdummy support, see wiki) For some reasons we've moved away (non blade related) from the blades, but not because blades don't work. We really liked them. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] annoying static when calling from legacy PBX - * ZAP interface
Hi all. I've got an Auerswald 4410USB ( http://www.auerswald.de/int/products/c4410usb.htm ) which I connected to my 2nd ZAP interface (s0 - Zap) via Crossoverr ISDN cable (which I crimped myself, I guess that's not the source of my trouble). Now what is annoying however, there is a very loud and very distinct static noise that is audible to the external party (outside of my * box within the telco network). The internal party does hear static, but MUCH more silent one. The * Box itself has 4 PCI slots, 3 of which are full: 10/100MBit-NIC, 1-port-ZAP, 1-port-ZAP. no VGA card, 1 Harddisk, nothing else. Now - what can be the issue here? Any help and pointers are very appreciated! -- best, Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Intel Blade Machine...
On Mon, Jul 04, 2005 at 02:15:36PM +0200, Andreas Sikkema wrote: Nahid Hossain wrote: I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? We've had Asterisk running on a blade for some time. Blades as such can be used but with a couple of restrictions: - There's probably no room for PCI cards, so no zap hardware - Check the kind of USB supported on the board (UHCI vs OHCI, for ztdummy support, see wiki) If you have no zaptel hardware and must rely on software you should use kernel 2.6's ztdummy, don't you? It is better, and also does not rely on USB. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7920
Betül Gözlükoğlu wrote: Thanks for link roland. The document mentions about firmware for Cisco 7920 ? Is it SIP firmware ? I asked the cisco seller for Sip firmware but they said sip firmware is unavailable for 7920? Would Creating files mentioned on the document be enough for configuration? Thanks again You will need to get the sccp channel software here: http://chan-sccp.berlios.de/ Download it and follow the instructions included. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Intel Blade Machine...
[EMAIL PROTECTED] wrote: We've had Asterisk running on a blade for some time. Blades as such can be used but with a couple of restrictions: - There's probably no room for PCI cards, so no zap hardware - Check the kind of USB supported on the board (UHCI vs OHCI, for ztdummy support, see wiki) If you have no zaptel hardware and must rely on software you should use kernel 2.6's ztdummy, don't you? It is better, and also does not rely on USB. Yes, true, but this entirely depends on how the blade is set up. We had no control over the distro installed on the blade. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] radius client for portaone with asterisk-1.0.9
hello i am trying to work with radiusclient form portaone. but i have some problems in installation. when i am trying to use example from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth error sip debug Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. BEGIN failed--compilation aborted at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. Jul 4 16:56:33 WARNING[5274]: app_dial.c:516 dial_exec: Dial argument takes format (technology1/number1technology2/number2...|optional timeout) Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Did the Broadvoice patch break asterisk ?
I think it might have. My understanding is BV does not require username and secret for incoming calls from BV to Asterisk, so a patch was written and then included in Asterisk to fix this. I have been testing for 4 weeks now, and have been shot down once in the digium bug tracker and pretty much blown off, but I'll try again here to see if I get anywhere. I and others noticed that we get pretty weird and annoying interactions between incoming Broadvoice calls and Asterisk. Specifically... The CDR is wrong, shows the wrong Broadvoice Number (the username in the trunk setup) The Asterisk system follows the wrong context for the trunk. After a LOT of testing what I seemed to have found is that Asterisk has been programmed to cache the information for the first incoming Broadvoice call, the number and the context. All subsequent calls from Broadvoice will use this number and context REGARDLESS of what the other Broadvoice trunks have been programmed with no matter how wrong the settings are for them, no errors, just chugs along using the info for the first trunk that rings in. So if the first trunk has a context that tells it to use the auto attendant, then ALL BV trunks will use the AA even if their context is to do something else. You can give all the other BV trunks a non-existent context and their is no error when they ring in, they just follow the valid context from the first trunk that rang in. The work aorund has been to use the /EXT option in the registration, but that is not a good fix, as other interactions are occuring as well. I have 14 BV trunks, and I have repeatedly proven that Asterisk will ignore ALL cutomizations for their username and context and instead will follow the settings for the first one that rings in following a reboot. Am I missing something here, or is this indeed a problem ? Our logs files and CDR are all screwed up and meaningless with respect to tracking back to the actual SIP CHANNEL activity as they always reflect the first trunk that rang in for all BV trunk activity. I am using the most current release of Asterisk. If I recall correctly (and I'm not a BV user), incoming sip calls to asterisk do not use all the parameters as one would expect. I think Olle posted something in the last month or two that such incoming calls attempt to find the first context that matches IP Address (or something like that) regardless of other parameters passed. The sip.conf.sample file includes the following comment: ; For incoming calls only. Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) You might try to find that post and review the exact sequence of parameters that asterisk uses for incoming sip calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions will not go to voicemail
I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial 710) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-5, SIP/710|30|tr) in new stack -- Called 710 -- SIP/710-4841 is ringing == Spawn extension (office, 710, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569-5' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' and the dialplan for that context is [office] exten = 710,1,NoOp(Dial 710) exten = 710,2,Dial(SIP/710,30,tr) exten = 710,3,Voicemail(u710) exten = 710,103,Voicemail(b710) Any ideas why I am not getting to the voicemail for that extension? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM01B card configuration
I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel = 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling requested on channel 4 is FXO Loopstart but line is in FXS Loopstart Signaling chan_zap.c:9927 setup_zap: Unable to register channel '4' Maybe I have something misconfigured. However, I triead all combinations and it doesn't seem to work. If I take away those lines. Asterisk comes up and in zaptel show command, I see the TDM400P card. Please help. Thanks What do you see listed in 'dmesg'? It should look something like: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) (The above example is from a TDM with four fxo modules installed.) What do you see listed in 'ztcfg -vv'? It should look something like: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) (Again, the above is from a TDM with four fxo modules, each configured for Kewlstart, not loopstart.) If you're not seeing entries similar to the above, I'd have to guess that changes made to /etc/zaptel.conf and /etc/asterisk/zapata.conf are probably out of sync. (Meaning changes have been made to one or the other without restarting asterisk AND zaptel.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7920
Joseph [EMAIL PROTECTED] : Would Creating files mentioned on the document be enough for configuration? You will need to get the sccp channel software here: http://chan-sccp.berlios.de/ Download it and follow the instructions included. Please keep in mind that is under development. The code is not at all tested for a production server. It will soon. Expect it to crash, random hangups, etc. btw it is working :-) Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions will not go to voicemail
I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial 710) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-5, SIP/710|30|tr) in new stack -- Called 710 -- SIP/710-4841 is ringing == Spawn extension (office, 710, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569-5' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' and the dialplan for that context is [office] exten = 710,1,NoOp(Dial 710) exten = 710,2,Dial(SIP/710,30,tr) exten = 710,3,Voicemail(u710) exten = 710,103,Voicemail(b710) Any ideas why I am not getting to the voicemail for that extension? What does the CLI look like on the pbx that has exten=710 defined? If you do a 'sip show peers' on that remote pbx, is 710 in the list? Is x710 on the same lan segment as the remote pbx? (If not, are there any nat boxes or firewalls involved?) Does a local call from some extension on that remote pbx to x710 on the same pbx work correctly with voicemail? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 12 seat call centre with Asterisk, VoIP only, UK - possible?
Talk to BT about getting an ISDN30 line put in... you'll get some sort of guaranteed quality and it'll be much better than a pure SIP solution. Talk to anyone APART from BT, their pricing is much more than OLOs etc. Who else can I order an ISDN30 line from in the UK?? I am looking for one at the moment. £300pm for a 2Mbs / sdsl / 1:1 contention / no router / seems to be the cheapest going rate. You shouldn't run anything important like that on an uncontended service as you are not sure of your bandwidth. I would put a pure internet solution (using ulaw/alaw) between a mobile and a landline due to variances in call quality. Especially if it is mostly inbound I would stick with ISDN carrier pre select. Thanks Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions will not go to voicemail
Chris Mason (Lists) wrote: -- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial 710) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-5, SIP/710|30|tr) in new stack -- Called 710 -- SIP/710-4841 is ringing == Spawn extension (office, 710, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569-5' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' My guess would be that system placing the IAX2 call has a shorter timeout than this one, so this one never gets a chance to go to voicemail. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp fon behind NAT gw
Am Mo, den 04.07.2005 schrieb Mathias Röhl um 13:40: ok, *done*, my fault, error in NAT configuration... regards mathias roehl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 licencing with asterisk, how does it work ??
Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? to whom ? how do I install them on asterisk etc ? thanks in advance , jl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 licencing with asterisk, how does it work ??
find it here: http://www.digium.com/index.php?menu=product_detailcategory=extrasprod uct=G729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty Sent: Monday, July 04, 2005 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] G729 licencing with asterisk, how does it work ?? Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? to whom ? how do I install them on asterisk etc ? thanks in advance , jl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Virtual Machine
Dear All : We are using [EMAIL PROTECTED] We did install [EMAIL PROTECTED] on a Virtual Machine .. All the SIP Calls are working fine .. But - We noticed that the codec on MeetMe Application is not working probably How can we solve this problem ??? Thanks ,, Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 licencing with asterisk, how does it work ??
Jean-Louis curty wrote: Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? On Digium's website. http://www.digium.com/index.php?menu=product_categorycategory=extras to whom ? Digium. how do I install them on asterisk etc ? Instructions are provided with your purchase. Installation is very easy provided you're used to unix / linux and terminal windows / consoles. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Virtual Machine
did you use the zaptel drivers? you need a timer interface for meetme application! use ztdummy! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed Farid Sent: Monday, July 04, 2005 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk on Virtual Machine Dear All : We are using [EMAIL PROTECTED] ... We did install [EMAIL PROTECTED] on a Virtual Machine .. All the SIP Calls are working fine .. But - We noticed that the codec on MeetMe Application is not working probably ... How can we solve this problem ??? Thanks ,, Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202 333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail = SMS
On 7/1/05, Mark Charlton [EMAIL PROTECTED] wrote: On 7/1/05, Wilson Pickett [EMAIL PROTECTED] wrote: I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. I have been fighting with the Bayham Systems FastSMS AGI script, and I re-wrote it as a stand alone Perl script. I am now calling it with the EXTERNNOTIFY option in the voicemail.conf file. It gets passed the context, extension and number of messages which I build into a text, and since they all go to the same location its no problem. I'm planning on using the extension info to open the mailbox, and read the text file for the latest message to pull out the caller for the text. I might also have an extension map in a text file so I can look up who to notify about a VM. I thought I had this one fixed, but now it doesn't seem to work. The solution worked for a day and sent the sms messages as voicemails came in. After that it didn't trigger the sms script at all. I can run it manually no problem, and the emails are still being sent for the voicemail. I updated asterisk to Asterisk CVS-Nv1-0-9-07/03/05-16:25:44, but the symptoms persist. Is there anyway I can debug the externnotify command? It doesn't repot anything to the CLI. I have tried writing to a log file in the script, but its not getting that far. I don't recall changing anything once I got it working. I didn't upgrade * until yesterday when it had been playing up for a few days. Any suggestions. Thanks agian Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telephoning Announcements -- Suggestions?
On Jul 2, 2005, at 11:35 PM, Jay Milk wrote: That's all doable. How many residents are you talking about? -- could take quite a while to call them all. Tell me about it -- we're doing it manually now! Considering you have outlay in hardware, phone-cost, utilities (a 100W computer draws $5-$10/month), consider fixing that well as someone suggested. Well, (no pun intended), not all of the problems are mechanical (although the current one is). Most of the time it is because some of the residents don't think about others until we remind them, and they have emptied the pressure tank and we all loose water pressure. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer using SIP clients
Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently using Asterisk CVS as of July 4th 2005. The only means of communication at the moment is the XTen X-Lite SIP Client, I already added the following entries to my sip.conf configuration file: [frank] canreinvite=no type=friend secret=frank username=frank nat=yes host=dynamic [test] canreinvite=no type=friend secret=test username=test nat=yes host=dynamic The SIP setup is working without a problem, the X-Lite application correctly registers the users and I can set up calls between them. I've also tested queues and they work without a problem, too. Next up is my extensions configuration, of which the interesting section now looks like this: [default] include = general ; longshot, added out of desparation include = parkedcalls ; longshot, added out of desparation include = featuremap ; longshot, added out of desparation exten = 800,1,Answer exten = 800,2,Dial(SIP/frank,20,tT) exten = 800,3 Hangup exten = 802,1,Answer exten = 802,2,Dial(SIP/test,20,tT) exten = 802,3 Hangup Notice the inclusion of several contexts that should or would have to be defined in the features configuration. My features.conf looks something like this, I trimmed the 'general' section for brevity: [general] ; (trimmed) default options [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer My testing scenario starts as follows: - log in both X-Lite SIP clients - from the 'test' phone, call extension 800 - on X-Lite client 'frank' accept the call - talk to eachother At this point I want to transfer to call to another extension, also defined in sip.conf but unlisted here. The problem is that nothing happens when I press the #1 or *2 keys in the 'frank' X-Lite client. I also tested these key combinations on the 'test' X-Lite client during the call, but that also had not effect. I searched the web and the mailing list archive for a solution, and if I recall correctly, someone stated that call transfer is only available for calls originating from the PSTN. Is this correct, also in regard of the current version of Asterisk? Has anyone got an idea how to get call transfer to work? One thing I tried was to change the DTMF settings in the clients, so they are sent in-band, but this also didn't help. Should I revert this option? Thanks in advance for your time and patience. Sincerely, Frank Schoep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail (gui vmail.cgi) patch
Hi, How could I change the defaultpermissions for voicemails? When I try to installthe patch mentionedat http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi, I get the following response: patch voicemail.patch patching file app_voicemail.cHunk #1 FAILED at 39.Hunk #2 FAILED at 119.Hunk #3 FAILED at 296.Hunk #4 FAILED at 1248.Hunk #5 FAILED at 1273.Hunk #6 FAILED at 1296.Hunk #7 FAILED at 1398.Hunk #8 FAILED at 1567.Hunk #9 FAILED at 1676.Hunk #10 FAILED at 3451.10 out of 10 hunks FAILED -- saving rejects to file app_voicemail.c.rej I'm afraid Idon't have previous experience in patchingso I don't know what's going on (Just trying to guess what It could be,toorecentversion of app_voicemail.c, wrong command parameters..) It would be very interesting open the permissions to user nobody. What I am doing at the moment isexecute a cron job with chown -R for user nobody over the voicemail directory. Thanks, Victor. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Join wav Files in Linux
Does anyone know how to join two .wav audio files via the command line in Linux for playback with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7920
On 15:15:19 July 04, 2005 [EMAIL PROTECTED] wrote: Joseph [EMAIL PROTECTED] : Would Creating files mentioned on the document be enough for configuration? You will need to get the sccp channel software here: http://chan-sccp.berlios.de/ Download it and follow the instructions included. Please keep in mind that is under development. The code is not at all tested for a production server. It will soon. Expect it to crash, random hangups, etc. btw it is working :-) You are far to decent, Sergio! chan-sccp from Sergio is working fine and undergoing heavy stress testing here. Until now, I just found smaller inconsistencies, no real nasty bugs. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM:***** SpamScore] [Asterisk-Users] Join wav Files in Linux
On Monday 04 July 2005 16:31, Kevin Kiely wrote: Does anyone know how to join two .wav audio files via the command line in Linux for playback with Asterisk? Kevin, you might want to try Sox, see http://sox.sf.net for more information. I'm not sure it can join or concatenate audio files, but I think it will. Sincerely, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer using SIP clients
Frank Schoep wrote: Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently using Asterisk CVS as of July 4th 2005. The only means of communication at the moment is the XTen X-Lite SIP Client, I already added the following entries to my sip.conf configuration file: [frank] canreinvite=no type=friend secret=frank username=frank nat=yes host=dynamic [test] canreinvite=no type=friend secret=test username=test nat=yes host=dynamic The SIP setup is working without a problem, the X-Lite application correctly registers the users and I can set up calls between them. I've also tested queues and they work without a problem, too. Next up is my extensions configuration, of which the interesting section now looks like this: [default] include = general ; longshot, added out of desparation include = parkedcalls ; longshot, added out of desparation include = featuremap ; longshot, added out of desparation exten = 800,1,Answer exten = 800,2,Dial(SIP/frank,20,tT) exten = 800,3 Hangup exten = 802,1,Answer exten = 802,2,Dial(SIP/test,20,tT) exten = 802,3 Hangup Notice the inclusion of several contexts that should or would have to be defined in the features configuration. My features.conf looks something like this, I trimmed the 'general' section for brevity: [general] ; (trimmed) default options [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer My testing scenario starts as follows: - log in both X-Lite SIP clients - from the 'test' phone, call extension 800 - on X-Lite client 'frank' accept the call - talk to eachother At this point I want to transfer to call to another extension, also defined in sip.conf but unlisted here. The problem is that nothing happens when I press the #1 or *2 keys in the 'frank' X-Lite client. I also tested these key combinations on the 'test' X-Lite client during the call, but that also had not effect. I searched the web and the mailing list archive for a solution, and if I recall correctly, someone stated that call transfer is only available for calls originating from the PSTN. Is this correct, also in regard of the current version of Asterisk? Has anyone got an idea how to get call transfer to work? One thing I tried was to change the DTMF settings in the clients, so they are sent in-band, but this also didn't help. Should I revert this option? Thanks in advance for your time and patience. Sincerely, Frank Schoep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I don't know if this will be of any help to you, but at least I can confirm problems with transfering calls with SIP agents. A little while ago we were having big problems getting transfers using DTMF to work. In that particular situation we were using a mix of only hard SIP devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both the stable version of asterisk and the CVS HEAD, but without results (but negative). In the end, we solved the problem by not using DTMF transfers at all, but by using the transfer capabilities of the SIP devices themselves (transfer for and hold buttons). These buttons did not appear to work (correctly) with the stable asterisk version we initially used (1.0.7), but with the CVS HEAD ( 29-MAY-2005) they appear to work just fine. I'm not familiar with soft SIP agents, so I don't know if the ones you use have such build-in transfer capabilities as their hardware counterparts like the BT101's and Snom190's have. I they do, you might wan't to give it a try. This is of course rather a workaround than a solution to your problem. E. Andriol -- --- HeuvelTop ICT Diensten v.o.f. --- There are management solutions to technical problems, but no technical solutions to management problems --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Join wav Files in Linux
On Mon, Jul 04, 2005 at 10:31:13AM -0400, Kevin Kiely wrote: Does anyone know how to join two .wav audio files via the command line in Linux for playback with Asterisk? install the package sox (should be part of most distros). sox infile1 [...] outfile1 e.g: sox in1.wav in2.wav in3.wav out.wav Sox generally tries to do the right thing with mixing inputs to outputs. This means that is all the inputs are exactly in the same format and the extensions are right, you should have no problems, but if you, you may need some extra hints using extra command-line switches. BTW: sox supports gsm as well. Should also support ogg/vorbis quite well. It generally supports decoding mp3-s, but at least on my system (Debian Sarge) this support is quite spotty. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P FXO PCI Card + Incoming Fax
Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial *97 to pickup voicemail buts says mypasswordincorrect
Not sure why I see *97 and *98 here, but I would check your dtmfmode= line in sip.conf. Often times, using rfc2833 works when inband or sip-info doesn't. See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Angus Comber |Sent: Monday, July 04, 2005 4:34 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Dial *97 to pickup voicemail |buts says mypasswordincorrect | | |I have found that if I dial from another extension *98 and |select extn 200 |and enter password 1234 it works. So is it something to do with |configuration on my IP Phone? It is a Grandstream GXP2000 running: |Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3 | |Anyone got any ideas? | |Angus | | | |- Original Message - |From: Angus Comber |To: asterisk-users@lists.digium.com |Sent: Monday, July 04, 2005 12:20 PM |Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my |passwordincorrect | | |Hello | |I am at extension 200 and I know there is a voicemail message |waiting. I |dial *97 and am prompted for the password. I enter 1234 which |I have set as |my voicemail password. What can I do to troubleshoot? | |Angus Comber |Itel Office Software Ltd |5 Enmore Gardens |London, SW14 8RF |Tel: 020 8878 7367 |Fax: 020 8876 7257 |Em: [EMAIL PROTECTED] |web: www.iteloffice.com | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
Paul Goodyear wrote: Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul. Yeah I recall there is a module for asterisk to do this. Search the list archives. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? I haven't tested the digium x100p for several months, but I believe it has the same issue as the TDM card relating to missed data frames across the pci bus. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
On 7/4/05, Rich Adamson [EMAIL PROTECTED] wrote: Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? I haven't tested the digium x100p for several months, but I believe it has the same issue as the TDM card relating to missed data frames across the pci bus. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Yes, using spandsp. My own experience has beent his works on receive 80% of the time. Some machines never are able to sync with it somehow. That's just my experience. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
Would it be possible to see your fax sections in your Extensions.conf file to see what you have there? This is a good place to start: http://scottstuff.net/blog/articles/category/Asterisk?page=4 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questions about real-time voicemail, foreign languages and voicemail folders...
On Sun, 2005-07-03 at 12:31 -0400, Leo Burd wrote: Hello there, I'm trying to configure my voicemail system and I have a couple of questions: * Is real-time voicemail already working? If so, where is it that I should specify the database name, user and password? Where can I get more information about the different options that exist and the different files that need to be changed? * Is it possible to setup the voicemail interface to speak in Spanish? * I've heard it is possible to create folders within mailboxes. Is there any documentation written about that? In fact, would anyone recommend a good reference about comedian mail? Thanks in advance, Everything you want is possible. Please read the Asterisk documentation at www.voip-info.org to set up your system. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Astmanproxy 1.1 now available!
Hey there folks -- I have been continuing development on the multi-threaded, c-based Asterisk Manager Proxy program, AstManProxy. I've incorporated several ideas I received at the recent Astricon Europe, including: - Supports proxying of multiple Asterisk servers at once - Abstracted, modular I/O handlers (implemented as shared objects) - Existing handlers: XML, Standard, CSV, HTTP One really cool feature that I'd like feedback and testing on is HTTP support. With this, you can POST or GET HTTP to the proxy and receive XML back, thus allowing a very simple (REST-like) web interface to the Asterisk Manager. Please download astmanproxy 1.1 and try it out: http://www.popvox.com/astmanproxy We are also putting together an 'astmanproxy' mailing list. If you would like to join, please e-mail me and I will include your name in the initial mailing list. Thanks, and for you fellow yanks out there, have a great holiday! Regards, Dave -- David C. Troy President/CEO popvox, LLC [EMAIL PROTECTED] Phone: +1-410-647-5812 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail (gui vmail.cgi) patch
HI, usually those errors arise when you try to use a patch with a different version. It happened to me when I tried to patch Asterisk 1.0.7 with a different version patch. Giorgio Incantalupo Victor Alvarez wrote: Hi, How could I change the default permissions for voicemails? When I try to install the patch mentioned at http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi, I get the following response: *patch voicemail.patch* patching file app_voicemail.c Hunk #1 FAILED at 39. Hunk #2 FAILED at 119. Hunk #3 FAILED at 296. Hunk #4 FAILED at 1248. Hunk #5 FAILED at 1273. Hunk #6 FAILED at 1296. Hunk #7 FAILED at 1398. Hunk #8 FAILED at 1567. Hunk #9 FAILED at 1676. Hunk #10 FAILED at 3451. 10 out of 10 hunks FAILED -- saving rejects to file app_voicemail.c.rej I'm afraid I don't have previous experience in patching so I don't know what's going on (Just trying to guess what It could be, too recent version of app_voicemail.c, wrong command parameters..) It would be very interesting open the permissions to user nobody. What I am doing at the moment is execute a cron job with chown -R for user nobody over the voicemail directory. Thanks, Victor. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution for Asterisk server use
On Sun, 3 Jul 2005, Subhi S Hashwa wrote: Telephony is a critical system to a business, if your phone system is down your business is as good as dead. If it costs me £600 for OS with support for 3 years, it's a price worth paying in the grand scale of things. You're buying Xeon server, Digium card, Digium license for G729 why not pay a small amount of money for peace of mind if the OS decides in the future it doesn't like your tie one day. Think of it as insurance. I'm assuming you are buying Asterisk Business Edition? If not, why not? Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: another database question
Yair,
When you have an older version you can try to use DBput/DBget (if still
working, because set will replace it in CVS)
Set(DB(CFIM/${CALLERIDNUM})=u${CALLERIDNUM})
will be;
DBput(CFIM/${CALLERIDNUM}=u${CALLERIDNUM})
Set(CFIM=${DB(CFIM/${ARG1})})
will be;
DBGet(${DB(CFIM/${ARG1})
normaly a database entry looks like:
CFIM/999 : 999
What is the line you use to fill the database with: /DB(CFIM/999) : 999 ?
What is the version of asterisk your machine runs?
Regards,
/* Ferdy */
http://asterisk.nsec.nl
info(AT)nsec(DOT)nl
Yair Hakak wrote:
hi ferdy,
i did check your first post to the list, and i really appreciate your help.
however, when i run your code i get an error because the set
application is not recognized - perhaps it is a CVS-head thing?
thanks,
yair
On 7/3/05, Ferdy Riphagen [EMAIL PROTECTED] wrote:
Yair,
Check my first post to the list, about your other question (call
forwarding, most basic case)
SetVar will be removed (I heard)
Greetz,
/* Ferdy */
Yair Hakak wrote:
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following statement:
exten = s,1,SetVar(CFIM=${DBGet(CFIM/${ARG1})})
i get the following:
Executing SetVar(Local/[EMAIL PROTECTED],2, CFIM=) in new stack
any ideas why the CFIM variable is not getting the 999 value?
thanks for any help,
yair
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[Asterisk-Users] Asterisk 1.0.9 and FreeTDS
Hi all, I have a working 1.0.7 installation and it is recording CDR to mysql. I am using FreeTDS on the system currently to access our MS SQL 2000 server for account verification, we may use it to store CDR records there in the future. I have decided to update the installation to 1.0.9. However, during make, I receive: make[1]: Entering directory `/usr/src/asterisk/cdr' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\1.0.9\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o cdr_tds.o cdr_tds.c cdr_tds.c: In function `mssql_connect': cdr_tds.c:415: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:415: error: (Each undeclared identifier is reported only once cdr_tds.c:415: error: for each function it appears in.) cdr_tds.c:415: error: `connection' undeclared (first use in this function) cdr_tds.c:437: error: too few arguments to function `tds_alloc_context' cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect' cdr_tds.c: At top level: cdr_tds.c:71: warning: `connect_time' defined but not used make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 I checked the version of FreeTDS, it is freetds v0.64.dev.20050704. (upon seeing on another post checking version of FreeTDS, I updated it to the most recent one) I checked, tds.h exists in /usr/local/include. Any ideas what is causing the error? Regards, Remzi Semsettin Turer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and IM again, want to develop a working hack
Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and found lots of wonderful proposals, which try to convince asking users, what needs to be done to support this well (multichannel, multiple phones per user, ...), mainly saying, that without very difficult reworking of internals, it would not be supported. What I really need is to hack it into chan_sip.c. I need the support of other channels and applications (f.e. MeetMe), but where I really care about presence and IM is SIP. So, any project, hack, patch, anything, that would allow me to go further with this would be greatly appreciated. I found this page in Russian: http://www.asterisk-support.ru/forums/development/53843189454 that somehow deals with the problem. I tried babelfish translation, (http://babelfish.altavista.com/babelfish/trurl_pagecontent?lp=ru_entrurl=http%3a%2f%2fwww.asterisk-support.ru%2fforums%2fdevelopment%2f53843189454) but I was not able to find out, if it really at least partially solves this problem, but as far as I understand it, Windows Messanger makes use of Subscribe/Notify, so this should be it. b.) Anyone has a partial solution using SER (which supports presence and IM) as a frontend, but routing all calls through Asterisk? Can this be done? I need the calls to go via Asterisk (I don't mean only sip notifications, but also the data, so I have canreinvite=no). So basically, SER would be a registrar proxy to Asterisk, which would do the authentication. The only thing, that SER would do would be to handle presence and IM and pass everything else on to Asterisk (as far as I know, SER can't pass traffic through it. I need the data to pass through the SIP server, since machines in my network topology don't see each other, it's a star with Asterisk in centre -- quite poetic indeed:). Any ideas, pointers to similiar configurations, ... are welcome. c.) If there is no solution to start with, is it possible to implement it only to chan_sip? I'm not familiar with Asterisk source code at all. Where are the places to look (in chan_sip.c) which are best to hook this code. Again, any code, hints, etc. about the structure of the source code are really welcome. Doing this in a clean way (although it's a hack) so it can be reused by community as much as possible is my intent. If anyone wants to help with the project by donating coder's time, mail me off the list. I hope I'll be able to support presence for hardphones and Xten's eyeBeam softphone in a few days with your help. Best wishes and thanks for any replies, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Did the Broadvoice patch break asterisk ?
If I recall correctly (and I'm not a BV user), incoming sip calls to asterisk do not use all the parameters as one would expect. I thinkOlle posted something in the last month or two that such incomingcalls attempt to find the first context that matches IP Address (orsomething like that) regardless of other parameters passed.The sip.conf.sample file includes the following comment:; For incoming calls only. Example: FWD (Free World Dialup); We match on IP address of the proxy for incoming calls; since we can not match on username (caller id)You might try to find that post and review the exact sequence ofparameters that asterisk uses for incoming sip calls. Yes, I know all of that, the problem is asterisk is NOT trying to match anything except the IP from Broadvoiice. So all calls from BV will be cached to the phone number and context of the first BV call. Asterisk will not look at the phone number or context of the other BV number when they come in. So... Since Asterisk IS GETTING the phone number and context for the first one that rings in, but IS NOT getting the phone number and context for subsequent BV calls and INSTEAD caches and uses the info from the first call, ASTERISK IS BROKEN. Think about it and I think anyone would agree this is not a good thing (can't send calls to correct destination based on context, can't bill the proper partry for their phone service). Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel and 2.6.13-rc1
Using a vanilla 2.6.13-rc1 kernel zaptel compiles but then when modprobing dmesg gives:- zaptel: Unknown symbol class_simple_device_add zaptel: Unknown symbol class_simple_destroy zaptel: Unknown symbol class_simple_device_remove zaptel: Unknown symbol class_simple_create Loads OK with 2.6.12.2 -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax DETECTION with CAPI
On Mon, 4 Jul 2005, sylvain garcia wrote: hi, I have dabian sarge with asterisk 1.0.7 and chan_capi 0.3.5 with AVM fritz card. I would like use detecion of fax, but it don't work. So, i would like know if it's possible to work fax detection with this card? And if it's possible how?? capiFax is integrated in chan_capi-cm (on sourceforge), but I cannot tell if the driver for the fritz card provides faxing... Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Did the Broadvoice patch break asterisk ?
Yes, I know all of that, the problem is asterisk is NOT trying to match anything except the IP from Broadvoiice. So all calls from BV will be cached to the phone number and context of the first BV call. Asterisk will not look at the phone number or context of the other BV number when they come in. So... Since Asterisk IS GETTING the phone number and context for the first one that rings in, but IS NOT getting the phone number and context for subsequent BV calls and INSTEAD caches and uses the info from the first call, ASTERISK IS BROKEN. Think about it and I think anyone would agree this is not a good thing (can't send calls to correct destination based on context, can't bill the proper partry for their phone service). Thanks Hi, is that the reason why asterisk doesn't work with multiple accounts at the same provider? I read somewhere about it and indeed have problems with calls from two nikotel accounts. If that's the case I'll vote for broken, too. Christian Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax out fails
on spamdsp page found thisIt seems possible for the libtiff library to fall over when handling some bad TIFF files. If spandsp is being used with Asterisk, this might bring the entire PBX down. So far only one person has reported this. Recent security update patches for libtiff 3.5.7, 3.6.0, and 3.6.1 hopefully correct this problem.verify your libtiff version.On 7/1/05, Bob Goddard [EMAIL PROTECTED] wrote: On Friday 01 Jul 2005 00:17, David Romero wrote: Are you sharing the IRQ of the zap card whit other device? some times when the zap card share IRQ whit other device spansdp fail.Turned out that while a fax may be a tiff file, it does not mean thata tiff file is a fax. The size of the generated tiff file was wrong. Interestingly though, when I try to fax out the PRI to one of ourown DDI's, that to say it come back in on the PRI, the fax softwarejust sits there looking stupid! On 6/30/05, Bob Goddard [EMAIL PROTECTED] wrote: I've a stock RH9 system with spandsp 0.18. Faxing out over a PRI to a USRobotics modem on a stock Suse9.3 system with hylafax fails with the following errors in the hylafax logs: Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 0, got 595, expected 1728 Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 1, got 595, expected 1728 Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 2, got 595, expected 1728 etc... I'm using the following call file as a test: Channel: Zap/g1/XX MaxRetries: 0 WaitTime: 20 Application: txfax Data: /root/t.tif|callerThe * console does not give any useful info even when the verbose setting is on max. The tiff file in question does not seem to be a problem is it is a 2 page file which is viewable just fine with xv. Does anyone have any clue as to what is wrong? It fails even if I set it up such that it dials out then back in on itself.___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM01B card configuration
The /var/log/messages lists: kernel: Module 0: Not installed kernel: Module 1: Not installed kernel: Module 2: Not installed Jul 3 22:21:10 kernel: Module 3: Installed -- AUTO FXO (FCC mode) kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) the ztcfg -vv: Zaptel Configuration == Channel map: Channel 04: FXS Loopstart (Default) (Slaves: 04) 1 channels configured. When you try to start asterisk. the following errors appear Jul 4 10:37:59 NOTICE[4015]: res_odbc.c:518 load_module: res_odbc loaded. .Jul 4 10:37:59 ERROR[4015]: chan_zap.c:6584 mkintf: Signalling requested on channel 4 is FXO Loopstart but line is in FXS Loopstart signalling Jul 4 10:37:59 ERROR[4015]: chan_zap.c:9927 setup_zap: Unable to register channel '4' Jul 4 10:37:59 WARNING[4015]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jul 4 10:37:59 WARNING[4015]: loader.c:523 load_modules: Loading module chan_zap.so failed! Any ideas? --- Rich Adamson [EMAIL PROTECTED] wrote: I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel = 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling requested on channel 4 is FXO Loopstart but line is in FXS Loopstart Signaling chan_zap.c:9927 setup_zap: Unable to register channel '4' Maybe I have something misconfigured. However, I triead all combinations and it doesn't seem to work. If I take away those lines. Asterisk comes up and in zaptel show command, I see the TDM400P card. Please help. Thanks What do you see listed in 'dmesg'? It should look something like: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) (The above example is from a TDM with four fxo modules installed.) What do you see listed in 'ztcfg -vv'? It should look something like: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) (Again, the above is from a TDM with four fxo modules, each configured for Kewlstart, not loopstart.) If you're not seeing entries similar to the above, I'd have to guess that changes made to /etc/zaptel.conf and /etc/asterisk/zapata.conf are probably out of sync. (Meaning changes have been made to one or the other without restarting asterisk AND zaptel.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! Well if you say it's registered, then packets are getting to asterisk and asterisk is accepting them, and you've allowed that SIP client. So... if you say there's absolutely NOTHING happening when the phone dials, then it sure seems like the phone is bad -- again, assuming no event whatsoever is happening when you dial. What else have you done to debug this? Have you registered the phone directly against another * box? Have you registered another phone against this * box? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco 5300
Hello Everyone, This is my first post, and this is my problem :-). I have a [EMAIL PROTECTED], work excellent (only internal users), but i need outbound calls. One person give me an access to his Cisco 5300 Media Gateway, he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the AMP config trunk ask me for a user and password, for the router, and i don't have it). Please, i'm newbie, and i don't know what type of trunk i need create, somebody can help me, with some manual or url. please! (Sorry my english please :-( ) Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird ring back
David Wilson wrote: I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. What phone are you using? A GXP-2000 or other model? I received notice of someone experiencing a sporadic 'ring' on his phone, but I'm not sure it was after haning up a call.. Do you use CDR logging (to mysql?), if yes, do you find anything in there about this call? Maybe try searching through the asterisk logfile for records of this call? Cheers, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware sizing
Hi all, I need some help/guidance on writing the specs needed on a project that will be scaling up to 10,000 users. I will have some T1's to provide PSTN connectivity, and all the users will be SIP and/or H323 phones. Services offered will include conferences, voicemail (20 megs per users), etc Should I use SER in front of asterisk to handle the SIP load ? I think I should put the voicemail server and conference server each on a seperate box. What kind of hardware would be preferred ? dual-Xeon or Quad Xeon ? What about redundancy ? What solutions can I look into for this ? Any help appreciated. Any link and/or documentation. I search with google but didn't find relevant informations. Maybe I just didn't put the right terms in the search box. thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gizmo: Skype done right?
I have a Gizmo account working perfectly in my Xten Eyebeam, so there should be no problem using it for Asterisk. You already have the username (1747...etc) and your password, the proxy is proxy01.sipphone.com (or you can sniff packets to see where SIP messages are being sent to). On 6/30/05, Robert Webb [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Thursday, June 30, 2005 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Gizmo: Skype done right? they claim to have a windows download but I can't get the program. also they give no instructions on how to get it connected to asterisk Which brings us to the question... Why is this being said to be good for Asterisk?? I did download it and load it on my computer. But there are NO options for connecting to anything or anyone else but a Gizmo account. So just how is this good for the open source VoIP community and Asterisk?? Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to know what happend after dial
when i dial an extension and the time on ring expiry how to know if called party is bussy or not answer. thanks-- David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to read dbm or voltage via ztmonitor ?
Hi the list, ztmonitor 3 -v start ztmonitor in graphical mode on Zaptel device #3. What is the correct syntax for dBm or voltage ? TIA Best Regards, Francois BERGERET, France. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to know what happend after dial
${DIALSTATUS} will tell you, also rtfm that will help you a lot.
The wiki is at: www.voip-info.org
Google is at: www.google.com
Browse this list: lists.digium.com
If you want to search the list with google, then type in
site:lists.digium.com when you enter your search terms on google.
On 7/4/05, David Romero [EMAIL PROTECTED] wrote:
when i dial an extension and the time on ring expiry how to know
if called party is bussy or not answer.
thanks
--
David Romero
##
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Re: [Asterisk-Users] X100P FXO PCI Card + Incoming Fax
El lun, 04-07-2005 a las 09:53, Paul Goodyear escribió:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
For me work fine this card, the spanDSP and the
Follow these steps:
/etc/asterisk/zapata.conf
faxdetect=incoming
/etc/asterisk/extensions.conf
exten = s,1,Wait,1
Then:
http://www.soft-switch.org/installing-spandsp.html
http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk
And here:
http://lists.digium.com/pipermail/asterisk-users/2005-April/103817.html
here, I made these changes:
/usr/bin/metasend -b -F $SENDER -t $RECIPIENT \
-s Fax de $FAXSENDER \
-S 1 \
-m 'text/plain' -f ${TMPFILE} -n \
-m 'application/pdf;name=fax'${FAXID}'.pdf' -f ${TMPFILE_A} \
-D 'PDF Fax Document'
(because, if the *.pdf file is too large, the metasend begin to split
it.)
Hope that this help you.
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
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[Asterisk-Users] Proper way to start * and load modules on a RedHat box
Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the alias stuff in modules.conf? Also how can I make the startup scripts appear in ntsysv? Even when I copy the scripts to rc.d they do not show up in ntsysv I tried loading the modules manually from rc.local but that doesn't work, even if I use delays. For some reason ztcfg doesn't work when run from rc.local and therefore asterisk fails to load. If I run ztcfg manually then ztcfg starts properly. Thanks for any hints / tips! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enable verbose output for TxFax/RxFax
Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes with a Philips fax machine. It seems that the fax machine doesn't recognize the carrier. How can I see the spandsp logs? I've enabled debug on the asterisk CLI, but I can't see any output while the txfax/rxfax application runs. Stefano Arata. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel and 2.6.13-rc1
Dave Cotton wrote: Using a vanilla 2.6.13-rc1 kernel zaptel compiles but then when modprobing dmesg gives:- zaptel: Unknown symbol class_simple_device_add zaptel: Unknown symbol class_simple_destroy zaptel: Unknown symbol class_simple_device_remove zaptel: Unknown symbol class_simple_create The sysfs interfaces have changed post-2.6.12, and the drivers will need updating. Please open a bug in Mantis about this issue so we don't forget it. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play message to callee beforeconnecttoincomingcall
You start to not make any sense, you posted a question like this: i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 any suggestions on how to implement this in an easy way? Using queues it all 5 things will happen. So now you adding some new stuff lets see. On 7/3/05, Roland Zagler [EMAIL PROTECTED] wrote: i had a look at the capabilities of queues and agents before and there are some missing points: -) the announcement has to be played to the caller until the callee answers (looped) I don't see why you can't use queues because of this, this is exactly what queues will do. -) the announcement has to start from the beginning (so i cannot use MoH) Again queues will do this for you. -) due to the missing ability of needed priorisation of agents, I thought you want to call one phone? Can you please explain what you mean with 'priorisation of agents'? I think it's implemented in HEAD. i cannot use the agents feature as implemented in asterisk Why not? for what you opened this thread it will do. Do you have something against queues? it looks like you are trying to avoid it. First you say realtime, which didn't tell me why you can't use queues. Now you come up with something else. We are trying to help you, but if you have a hard time taking help don't ask for help. On 7/3/05, Roland Zagler [EMAIL PROTECTED] wrote: i had a look at the capabilities of queues and agents before and there are some missing points: -) the announcement has to be played to the caller until the callee answers (looped) -) the announcement has to start from the beginning (so i cannot use MoH) -) due to the missing ability of needed priorisation of agents, i cannot use the agents feature as implemented in asterisk roland -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Monday, July 04, 2005 1:11 AM To: Roland Zagler Subject: Re: [Asterisk-Users] play message to callee beforeconnecttoincomingcall I don't see why this doesn't work with realtime. The same it works with .conf files On 7/3/05, Roland Zagler [EMAIL PROTECTED] wrote: Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, July 03, 2005 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee beforeconnecttoincomingcall I beleive queues will do it all for you. http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf Use this part to play the sound to the callers: ;queue-youarenext = queue-youarenext ; (You are now first in line.) and use: ;announce = queue-markq To announce what you want to the callee Hope this helps ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: another database question
hi ferdy,
again, thanks for all your help. I will try this and report back.
as for your questions:
1. my version is from stable, Asterisk CVS-v1-0-01/22/05-12:27:04
2. the line used that gets this database result is:
exten = 154,2,DBPut(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
which is, of course, wrong. i'll fix that and i'll let you know how
everything works.
thanks again for all the help,
yair
On 7/4/05, Ferdy Riphagen [EMAIL PROTECTED] wrote:
Yair,
When you have an older version you can try to use DBput/DBget (if still
working, because set will replace it in CVS)
Set(DB(CFIM/${CALLERIDNUM})=u${CALLERIDNUM})
will be;
DBput(CFIM/${CALLERIDNUM}=u${CALLERIDNUM})
Set(CFIM=${DB(CFIM/${ARG1})})
will be;
DBGet(${DB(CFIM/${ARG1})
normaly a database entry looks like:
CFIM/999 : 999
What is the line you use to fill the database with: /DB(CFIM/999) : 999 ?
What is the version of asterisk your machine runs?
Regards,
/* Ferdy */
http://asterisk.nsec.nl
info(AT)nsec(DOT)nl
Yair Hakak wrote:
hi ferdy,
i did check your first post to the list, and i really appreciate your help.
however, when i run your code i get an error because the set
application is not recognized - perhaps it is a CVS-head thing?
thanks,
yair
On 7/3/05, Ferdy Riphagen [EMAIL PROTECTED] wrote:
Yair,
Check my first post to the list, about your other question (call
forwarding, most basic case)
SetVar will be removed (I heard)
Greetz,
/* Ferdy */
Yair Hakak wrote:
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following statement:
exten = s,1,SetVar(CFIM=${DBGet(CFIM/${ARG1})})
i get the following:
Executing SetVar(Local/[EMAIL PROTECTED],2, CFIM=) in new stack
any ideas why the CFIM variable is not getting the 999 value?
thanks for any help,
yair
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Re: [Asterisk-Users] Gizmo: Skype done right?
I think they were hoping that the client would connect to Asterisk, which makes it kinda useless, really.. But connecting Asterisk to the Gizmo network is handy. -- Dana On 7/4/05, Adrian A [EMAIL PROTECTED] wrote: I have a Gizmo account working perfectly in my Xten Eyebeam, so there should be no problem using it for Asterisk. You already have the username (1747...etc) and your password, the proxy is proxy01.sipphone.com (or you can sniff packets to see where SIP messages are being sent to). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.0.9 and FreeTDS
In article [EMAIL PROTECTED], Remzi Semsettin Turer [EMAIL PROTECTED] wrote: Hi all, I have a working 1.0.7 installation and it is recording CDR to mysql. I am using FreeTDS on the system currently to access our MS SQL 2000 server for account verification, we may use it to store CDR records there in the future. I have decided to update the installation to 1.0.9. However, during make, I receive: [...various errors...] I checked the version of FreeTDS, it is freetds v0.64.dev.20050704. (upon seeing on another post checking version of FreeTDS, I updated it to the most recent one) I checked, tds.h exists in /usr/local/include. Any ideas what is causing the error? Updating FreeTDS was the problem. The following text is from asterisk/doc/README.tds: --- PLEASE NOTE The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. The cdr_tds module is known to work with FreeTDS version 0.62.1; it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug fix releases. The cdr_tds module uses the raw libtds API of FreeTDS. It appears that from 0.63 onwards, this is not considered a published API of FreeTDS and is subject to change without notice. Between 0.62.x and 0.63 of FreeTDS, many incompatible changes have been made to the libtds API. For newer versions of FreeTDS, it is recommended that you use the ODBC driver. --- You will either have to downgrade to 0.62.x of FreeTDS or change to ODBC. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
