Re: [Asterisk-Users] software to do sip stress tests
On Tue, Jun 20, 2006 at 01:45:44AM +0100, [EMAIL PROTECTED] wrote: Hi, I want to make some stress tests on two machines were I configured different implementations of open source sip servers. I'm thinking about making some graphics like CPU and memory usage extracted by SNMP while flooding my servers of sip calls. Does anybody know some good software to do that? On Debian: apt-get install mumin . Or a simple cron to collect some stats and later graph them with gnuplot or your favorite spreadsheet. sar can be handy. So is ps. Or do your own measurements. One thing to note: if Asterisk is highly stressed on CPU and runs in real-time scheduling priority (-p), any other process attempting to meassure data at that point will give slightly(?) wrong stats. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] finding mac addresses
On Mon, Jun 19, 2006 at 12:21:32PM -0800, Michael Wallette wrote: Sure--an nmap (http://www.insecure.org) ping scan will show this. For example, on my network, I have an DHCP-addressed Iaxy that usually camps out on 192.168.1.130. Running a ping scan with nmap returns the following: $sudo nmap -sP -v -v 192.168.1.130 Password: Starting Nmap 4.01 ( http://www.insecure.org/nmap/ ) at 2006-06-19 12:13 AKDT Initiating ARP Ping Scan against 192.168.1.130 [1 port] at 12:13 The ARP Ping Scan took 0.02s to scan 1 total hosts. DNS resolution of 1 IPs took 0.03s. Mode: Async [#: 3, OK: 0, NX: 1, DR: 0, SF: 0, TR: 1, CN: 0] Host 192.168.1.130 appears to be up. MAC Address: 00:03:64:00:15:61 (Scenix Semiconductor) Nmap finished: 1 IP address (1 host up) scanned in 0.758 seconds Raw packets sent: 1 (42B) | Rcvd: 1 (42B) $ While I don't yet have any VoIP phones on this network to test, I imagine nmap would find VoIP phones, as well. HTH! ping(8) is known to be SUID root on most systems, and thus no need for nmap. But if we're into installing extra tools that need to run as root, grab arping from your nearby distro repo. OTOH, nmap could be used to scan the whole network: nmap -sP 192.168.1.1-254 After which the arp table will be filled... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.9 cli -x doesn't flush?
Bryan == Bryan Field-Elliot [EMAIL PROTECTED] writes: Bryan We have a script which executes asterisk -n -r -x . Bryan With prior versions of Asterisk this worked fine, but having Bryan just upgraded to 1.2.9, we are finding that if the output is Bryan lengthy, then Asterisk seems to terminate before fully I've got an error like this and found http://bugs.digium.com/view.php?id=7326 May be that patch helps you. -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video phones probem
Hi all, I'm testing video phones with asterisk for the first time. Voice calls goes fine. I have problems with video session. Advices needed! here is asterisk log: Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp: Unknown SDP media type in offer: video 6072 RTP/AVP 34 here is sip.conf [minkpr] type=friend context=bandymas videosupport=yes secret=minkpr language=us host=dynamic nat=no dtmfmode=inband allow=all other video user looks the same. thanks in advance, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge
On Mon, Jun 19, 2006 at 04:59:53PM -0500, Mark W. Stoddard wrote: As far as hardware is concerned, I am using the following: * Dell Poweredge 2850 * 2GB RAM * 2x 73GB 10,000 SCSI drives mirrored * 1x Intel Xeon at 3.8GHz * 1x Digium TDM2400P Requires zaptel 1.2, IIRC. * Dual redundant power supplies. (is saying dual redundant redundant?) * Stock cooling * UPS I was curious what version Debian testing is up to, apparently 1.2.7. I must have been living on Mars to have missed that. Testing is a moving target. Aimed at starting to freeze well after 1.4 will be released. Hence if you use Testing, be prepared not only for some base components of your system to be upgraded, but also Asterisk. Frankly I would be recommend agaist using Testing (or Fedora Core, for quite similar reasons) for production system: you'll be forced to upgrade in order to get bugfixes. Great for a developemnt enviironment. Not so for production. I'll attempt an upgrade from stable to testing on a testing machine (might even give it a try using Xen). Xen is indeed one of the things that is much better in Etch. If the upgrade goes well, I'll consider upgrading the production system to testing, or at least the Asterisk packages and dependencies thereof. I believe that Debian testing is scheduled to go stable this in a few (6?) months, so it will be a good idea to at least see what I'm up against. If that is indeed your schedule, then perhaps Testing is better than Stable+backports. How long does it take to restart Asterisk? I know there is a way to start Asterisk so that if it goes down, it comes back up immediately, I believe you refer to safe_asterisk . Frankly I don't trust it. Find the problem that takes asterisk down and solve it. Asterisk should not crash. In most cases it doesn't. safe_asterisk will just hide crashes. Use an external service watch dog to report problems. safe_asterisk just complicates the starting and stopping of asterisk. that could be part of a solution right there. If the downtime is a second or two every few days, that's still adequate uptime for a commercial phone system. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How would you tet a FastAGI script
Hi,I would to develop my first FastAGI script.I would like to test it independently from Asterisk for the sake of simplicity.Which linux (or cygwin) tool is the best for that ?Using this tool, I will open a FastAGI connection, throw data in and read data from. With AGI script, echo or cat commands are enough.But what are the simplest one with FastAGI ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call limit function on sip channel to external pop
Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new channel, it seems to work, but putting the call_limit on an existing sip channel going out, it doesn't do anything. Anyone already had such an issue, or anyone knowing the best config for limiting outgoing sip channels to external sip providers? It's kind of urgent... Thanks in advance, and keep up the good work! Bram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
On 6/19/06, Remco Barendse [EMAIL PROTECTED] wrote: found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile again it is required to change KSRC=/usr/src/linux/ to KSRC=/usr/src/linux-2.6/ I wonder why neither florz nor kapejod fixes these problems (several modules do not compile). This is a distribution specific issue, so will probably never be patched. /usr/src/linux is the traditional location for the current kernel source, although /lib/modules/kver/build/ usually contains everything needed for a module build for 2.6 kernels and up. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call limit function on sip channel to external pop
On Tue, 2006-06-20 at 09:20 +0200, bram kortleven wrote: Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new channel, it seems to work, but putting the call_limit on an existing sip channel going out, it doesn't do anything. Anyone already had such an issue, or anyone knowing the best config for limiting outgoing sip channels to external sip providers? Previous answers to similar questions usually point to: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] show queue ... Invalid
Kevin == Kevin P Fleming [EMAIL PROTECTED] writes: What does it mean? Why is it Invalid? BTW, reload command fixes it, so the member receives queue calls. Kevin channel in logger.conf and then try this again. You should see Kevin a message from chan_sip saying something like Checking Kevin devicestate for ... and the peername... we need to see what Kevin that message says. I've done it. Just after start I don't see any chan_sip.c: Checking device state for messages ann all of queue's members is Invalid. But after reload command I see: Jun 20 13:31:55 DEBUG[52930] chan_sip.c: Checking device state for peer agat2 Jun 20 13:31:55 DEBUG[52930] chan_sip.c: Checking device state for peer agat2 Jun 20 13:31:55 DEBUG[52930] chan_sip.c: Checking device state for peer agat2 and queue's members can receives calls. Thank you! -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voiceone?
Hi! anyone from here, who uses voiceone as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help from here. That's all, thank you, Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ooh323 issues
Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial(SIP/yyy-2965, OOH323/[EMAIL PROTECTED]) in new stack --- ooh323_request - data [EMAIL PROTECTED] format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- [EMAIL PROTECTED] --- onNewCallCreated ooh323c_o_22 --- find_call +++ find_call setting callid number 203 Outgoing call xxx(ooh323c_o_22) - Codec prefs - (gsm|ulaw|g723) Adding capabilities to call(outgoing, ooh323c_o_22) Adding gsm capability to call(outgoing, ooh323c_o_22) Adding g711 ulaw capability to call(outgoing, ooh323c_o_22) Adding g7231 capability to call (outgoing, ooh323c_o_22) --- configure_local_rtp +++ configure_local_rtp +++ onNewCallCreated ooh323c_o_22 +++ ooh323_call -- Called [EMAIL PROTECTED] --- onCallEstablished ooh323c_o_22 --- find_call +++ find_call +++ onCallEstablished ooh323c_o_22 -- OOH323/xxx-a6f1 answered SIP/yyy-2965 -- Attempting native bridge of SIP/yyy-2965 and OOH323/xxx-a6f1 --- onCallCleared ooh323c_o_22 --- find_call +++ find_call --- ooh323_hangup hanging xxx +++ ooh323_hangup == Spawn extension (internal, 00263203, 1) exited non-zero on 'SIP/yyy-2965' --- ooh323_destroy Destroying xxx +++ ooh323_destroy When calling from the H.323 box to my Asterisk server, my SIP phone rings, and I get a ringing signal from the H.323 server, but when the SIP phone is answered, it goes dead with the following error message: --- onNewCallCreated ooh323c_10 +++ onNewCallCreated ooh323c_10 --- ooh323_onReceivedSetup ooh323c_10 --- find_user +++ find_user Adding capabilities to call(incoming, ooh323c_10) Adding gsm capability to call(incoming, ooh323c_10) Adding g711 ulaw capability to call(incoming, ooh323c_10) Adding g7231 capability to call (incoming, ooh323c_10) --- configure_local_rtp +++ configure_local_rtp +++ ooh323_onReceivedSetup - Determined context internal, extension 203 --- onAlerting ooh323c_10 --- find_call +++ find_call +++ onAlerting ooh323c_10 -- Executing Dial(OOH323/Customer-7849, SIP/yyy) in new stack -- Called yyy -- SIP/yyy-8a35 is ringing - ooh323_indicate 3 on call ooh323c_10 ooh323_indicate 3 on ooh323c_10 -- SIP/yyy-8a35 answered OOH323/Customer-7849 - ooh323_indicate -1 on call ooh323c_10 Jun 20 12:00:43 WARNING[18607]: src/chan_h323.c:951 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_10 ooh323_indicate -1 on ooh323c_10 --- ooh323_answer +++ ooh323_answer -- Attempting native bridge of OOH323/Customer-7849 and SIP/yyy-8a35 --- onCallEstablished ooh323c_10 --- find_call +++ find_call +++ onCallEstablished ooh323c_10 --- onCallCleared ooh323c_10 --- find_call +++ find_call == Spawn extension (internal, 203, 1) exited non-zero on 'OOH323/Customer-7849' --- ooh323_hangup hanging Customer +++ ooh323_hangup --- ooh323_destroy Destroying Customer +++ ooh323_destroy I've seen a couple of threads about this on the web, pointing toward codec mismatches, e.t.c. I've toggled the various codecs on the H.323 server and Asterisk, with no luck. I'm running Asterisk 1.2.9.1 and Add-Ons 1.2.3. All help appreciated. Cheers, Mark. pgpJyTacKieBE.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which is the best user GUI ?
Hi,I would like to customise an end user application like Centiles's callpad software ( http://www.centile.com/solutions-applications-callpad.php ).Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations.Are you aware of any software that could be used for this ? I've read www.voip-info.org User interfaces section ( http://www.voip-info.org/wiki/view/Asterisk+GUI).23 softwares are listed. Which one is your favorite for that ? Why ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newest Asterisk doesn't compile
Hi, I compiled 1.2.7 no problem, however with 1.2.9.1 I'm getting this: chan_zap.c: In function `pri_dchannel': chan_zap.c:9038: error: structure has no member named `call' make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/root/asterisk/20-jun-2006-upgrade/asterisk-1.2.9.1/channels' make: *** [subdirs] Error 1 Got all the newest of everything drwxr-xr-x 25 1000 1000 4096 Jun 20 06:03 asterisk-1.2.9.1 -rw-r--r-- 1 root root 10568287 Jun 6 12:38 asterisk-1.2.9.1.tar.gz drwxr-xr-x 7 1000 1000 4096 Jun 1 13:08 asterisk-addons-1.2.3 -rw-r--r-- 1 root root 750973 Jun 1 18:55 asterisk-addons-1.2.3.tar.gz -rw-r--r-- 1 root root 48 Jun 16 16:50 README-1st drwxr-xr-x 6 1000 100012288 Jun 20 06:00 zaptel-1.2.6 -rw-r--r-- 1 root root 676658 May 30 18:50 zaptel-1.2.6.tar.gz Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Newest Asterisk doesn't compile
AHHHA! I didn't update my libpri! On 6/20/06, Matt [EMAIL PROTECTED] wrote: Hi, I compiled 1.2.7 no problem, however with 1.2.9.1 I'm getting this: chan_zap.c: In function `pri_dchannel': chan_zap.c:9038: error: structure has no member named `call' make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/root/asterisk/20-jun-2006-upgrade/asterisk-1.2.9.1/channels' make: *** [subdirs] Error 1 Got all the newest of everything drwxr-xr-x 25 1000 1000 4096 Jun 20 06:03 asterisk-1.2.9.1 -rw-r--r-- 1 root root 10568287 Jun 6 12:38 asterisk-1.2.9.1.tar.gz drwxr-xr-x 7 1000 1000 4096 Jun 1 13:08 asterisk-addons-1.2.3 -rw-r--r-- 1 root root 750973 Jun 1 18:55 asterisk-addons-1.2.3.tar.gz -rw-r--r-- 1 root root 48 Jun 16 16:50 README-1st drwxr-xr-x 6 1000 100012288 Jun 20 06:00 zaptel-1.2.6 -rw-r--r-- 1 root root 676658 May 30 18:50 zaptel-1.2.6.tar.gz Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hitting * in a queue call hangs up?
It seems 1.2.9.1 does not correct this behavior... can I correct it somehow? On 6/12/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - BJ Weschke [EMAIL PROTECTED] wrote: This was a hardcoded feature in Asterisk 1.2.X versions. It's now an optional feature in /trunk and will be going forward. And this is only true for queue members that are chan_agent agents. If you don't use chan_agent, you won't see this behavior either. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phones probem
20 jun 2006 kl. 08.51 skrev Mindaugas Kuprys: Hi all, I'm testing video phones with asterisk for the first time. Voice calls goes fine. I have problems with video session. Advices needed! here is asterisk log: Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp: Unknown SDP media type in offer: video 6072 RTP/AVP 34 I would like to see a complete SIP INVITE from this phone and get more information about the type of phone. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which is the best user GUI ?
Is Centile a solution built ontop of Asterisk? It looks similar according to their feature list. http://www.centile.com/solutions-intraswitch-platform-systemmanagement.php and http://www.centile.com/solutions-intraswitch-platform-advancedfeatures.php On 6/20/06, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to customise an end user application like Centiles's callpad software ( http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ? I've read www.voip-info.org User interfaces section ( http://www.voip-info.org/wiki/view/Asterisk+GUI). 23 softwares are listed. Which one is your favorite for that ? Why ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
On Tue, Jun 20, 2006 at 09:30:38AM +0100, Steve Davies wrote: On 6/19/06, Remco Barendse [EMAIL PROTECTED] wrote: found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile again it is required to change KSRC=/usr/src/linux/ to KSRC=/usr/src/linux-2.6/ I wonder why neither florz nor kapejod fixes these problems (several modules do not compile). This is a distribution specific issue, so will probably never be patched. /usr/src/linux is the traditional location for the current kernel source, although /lib/modules/kver/build/ usually contains everything needed for a module build for 2.6 kernels and up. Actually, the first place that zaptel checks is /lib/modules/kver/build (a symlink, not a directory). Debian, RH, and probably some other distros generate that link. I'm not sure if a normal kernel tree 'make install' does it as well. So I would not call that distro specific. Florz's patch aims at correcting code, and not build system. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager DBDel action
Hi list, is there a possibility to delete a key from the astdb through the manager interface? I managed to put and to get a key but I do not know how to delete an entry. The problem is that I want to use the manager interface because I can communicate remotely with my * this way. TIA, Christophorus begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voiceone?
Neil, I have not tried it yet, but I wanted to say this to those that don't realize it: VoiceOne is GPL http://www.voiceone.it/documentation/licence/ I just thought that was interesting... it doesn't look like it from the first look. On 6/20/06, Neil Adona [EMAIL PROTECTED] wrote: Hi! anyone from here, who uses voiceone as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help from here. That's all, thank you, Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fail to make call
Hi I have the following configuration | UA1 --|-- asterisk1 ---+ UA2 --|-- asterisk2 ---+ DB UA3 --|-- asterisk3 ---+ UA4 --|-- asterisk4 ---+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register infomration. UA1 can make call to UA2,UA3 and UA4. UA2 can make call to UA1, UA3 but not UA4. UA3 can make call to UA4 but not UA1, UA2 UA4 failed to make call to all UA. From the CLI and log below, asterisk shows it can't create channel. As I expect, all UA should able to find each other. However, some of them are failed to find others. I have no idea why they can't find each other well. Is it the configuration problem? Anyone can help? -- Executing Dial(SIP/871966629896-5373, SIP/871966760539|15) Jun 20 17:42:16 NOTICE[23355]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/871966629896-5373, ) NOTICE[28269] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Softphone on Thinclient?
Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer call via AMI or dialplan
Thanks for all the help so far on this, but I was wondering if there was a way of simulating an attended transfer from the AMI or dialplan ? Julian. Moises Silva wrote: Piece of cake Julian: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Regards On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Softphone on Thinclient?
Hi Steve, We are running X-Lite on Wyse V90 terminals. They have Windows XP Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the onboard audio chip is very poor on voice quality. I guess X-Lite has Windows CE version. Check on www.counterpath.com. Idris -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Softphone on Thinclient? Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I have no
availability for high speed internet at my house to place a VoIP call over.
So, I have a standard phone plugged into the PAP2, The PAP2 plugs into the
network at my house to which the asterisk box is also connected, the
asterisk box has an FXO card that has the PSTN line plugged into it, this is
where the ZAP channel comes in. when i dial a local number asterisk simple
dials the number out the pstn line. If i dial a long distance number, the *
box dials a local phone number that I have through my VoIP provider which is
answered by an * box that I have at a different location using a line in
extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was a
little clearer this time and sorry for the confusion.
John M
On Monday June 19 2006 11:22 pm, Mike Fedyk wrote:
this does not make any sense.
How do you dial to a service provider from your * box? Does it use PPP
and IP? And then you connect to another * box that is on a cable
connection that receives the call over IP and then dials out to a voip
provider? How do any fxo devices come into this picture? How does a
zap channel come into this picture?
John Millican wrote:
Doug,
The interface that i dial to is at my Service provider and am not sure
what they are using. I dial out of my * box to a service provider number
which is answerd by an asterisk box that I have at another location on a
high speed cable connection, that box then dials the numberI ultimately
want to reach. I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfer call via AMI or dialplan
Check features.conf. If not uncomment the atxfer line and assign a key combination (Default is *2). Then use t and T switches in Dial command. Finally restart asterisk service. -Original Message- From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfer call via AMI or dialplan Thanks for all the help so far on this, but I was wondering if there was a way of simulating an attended transfer from the AMI or dialplan ? Julian. Moises Silva wrote: Piece of cake Julian: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action +Redirect Regards On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug in asterisk static realtime?
Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be stripped off by asterisk itself!! It should be easy to modify the script, but the problem would remain. Should it be filed as an Asterisk bug? -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working with Asterisk and SIP? Register for the Asterisk SIP Master class!
Want to become an Asterisk SIPmaster? Register for the Asterisk SIP Master Class, taking place in Chicago, IL, USA July 10-14 organized by Edvina in partnership with Digium. We're developing this new training now, creating labs with Asterisk and SIP express router, NAT traversals, realtime and much, much more. Learn more here: http://edvina.net/training/sipmasterclass/ and register today! Questions? E-mail [EMAIL PROTECTED] today! It's going to be a fun and very educational week in Chicago. We only have 15 seats for this class, so make sure you register quickly. See you in Chicago! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bug in asterisk static realtime?
20 jun 2006 kl. 13.33 skrev Andrea Spadaccini: Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be stripped off by asterisk itself!! It should be easy to modify the script, but the problem would remain. Should it be filed as an Asterisk bug? A semicolon in realtime separates multiple values, it is *not* used as a comment. So you should fix your script. regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE420P/TE415P?
Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels. Does anyone know when thease will be released and what they will cost when released? Thanks! http://pressroom.pulvermedia.com/digium/pr.php#0314c Regards, jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) On 6/20/06, Idris AVCI [EMAIL PROTECTED] wrote: Hi Steve, We are running X-Lite on Wyse V90 terminals. They have Windows XP Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the onboard audio chip is very poor on voice quality. I guess X-Lite has Windows CE version. Check on www.counterpath.com. Idris -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Softphone on Thinclient? Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk h323
Hi Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works . Please if u know send me how to active h323 protocol or the conversion protocol Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phones probem
Video started to work. Now intresting thing is that video size is half reduced than calling directly from phone to phone. Phones: Tatung tia-8800. I have attached sip messages. that else might be important..? one of phones is behind nat. mindaugas Olle E Johansson wrote: 20 jun 2006 kl. 08.51 skrev Mindaugas Kuprys: Hi all, I'm testing video phones with asterisk for the first time. Voice calls goes fine. I have problems with video session. Advices needed! here is asterisk log: Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp: Unknown SDP media type in offer: video 6072 RTP/AVP 34 I would like to see a complete SIP INVITE from this phone and get more information about the type of phone. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users interface: eth0 (217.9.241.224/255.255.255.224) filter: (ip) and ( port 5060 ) # U 217.9.240.114:5060 - 217.9.241.227:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. From: Giedriussip:[EMAIL PROTECTED];tag=0-13c4-a0-2559e-a2a6e31. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE. Via: SIP/2.0/UDP 192.168.144.161:5060;rport;branch=z9hG4bK-a0-2559e-6c8a0916. Allow: INVITE,ACK,CANCEL,BYE,OPTIONS. User-Agent: TTIC Video phone. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:5060. Content-Type: application/SDP. Content-Length: 306. . v=0. o=Client 2890844526 2890844526 IN IP4 192.168.144.161. s=TTIC SIP Video Phone. c=IN IP4 192.168.144.161. t=0 0. m=audio 6070 RTP/AVP 0 97. a=rtpmap:0 PCMU/8000. a=rtpmap:97 telephone-event/8000. m=video 6072 RTP/AVP 34. b=AS:768. a=fmtp:34 CIF=1 QCIF=2 MaxBR=7680. a=sendrecv. a=rtpmap:34 H263/9. # U 217.9.241.227:5060 - 217.9.240.114:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.144.161:5060;branch=z9hG4bK-a0-2559e-6c8a0916;received=217.9.240.114;rport=5060. From: Giedriussip:[EMAIL PROTECTED];tag=0-13c4-a0-2559e-a2a6e31. To: sip:[EMAIL PROTECTED];tag=as3df9a2b1. Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: sip:[EMAIL PROTECTED]. Proxy-Authenticate: Digest realm=asterisk, nonce=12889af6. Content-Length: 0. . # U 217.9.240.114:5060 - 217.9.241.227:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0. From: Giedriussip:[EMAIL PROTECTED];tag=0-13c4-a0-2559e-a2a6e31. To: sip:[EMAIL PROTECTED];tag=as3df9a2b1. Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK. Via: SIP/2.0/UDP 192.168.144.161:5060;rport;branch=z9hG4bK-a0-2559e-6c8a0916. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:5060. Content-Length: 0. . # U 217.9.240.114:5060 - 217.9.241.227:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. From: Giedriussip:[EMAIL PROTECTED];tag=0-13c4-a0-2559e-a2a6e31. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE. Via: SIP/2.0/UDP 192.168.144.161:5060;rport;branch=z9hG4bK-a1-255ee-4ddd0c84. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:5060. Proxy-Authorization: Digest username=giedrius, realm=asterisk, nonce=12889af6, uri=sip:[EMAIL PROTECTED], response=1d713c7e54f91434a36a5cac26ded396, algorithm=MD5. Content-Type: application/SDP. Content-Length: 306. . v=0. o=Client 2890844526 2890844526 IN IP4 192.168.144.161. s=TTIC SIP Video Phone. c=IN IP4 192.168.144.161. t=0 0. m=audio 6070 RTP/AVP 0 97. a=rtpmap:0 PCMU/8000. a=rtpmap:97 telephone-event/8000. m=video 6072 RTP/AVP 34. b=AS:768. a=fmtp:34 CIF=1 QCIF=2 MaxBR=7680. a=sendrecv. a=rtpmap:34 H263/9. # U 217.9.241.227:5060 - 217.9.240.114:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.144.161:5060;branch=z9hG4bK-a1-255ee-4ddd0c84;received=217.9.240.114;rport=5060. From: Giedriussip:[EMAIL PROTECTED];tag=0-13c4-a0-2559e-a2a6e31. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . # U 217.9.241.227:5060 - 217.9.241.234:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0. Via: SIP/2.0/UDP 217.9.241.227:5060;branch=z9hG4bK198aadad;rport. From: Giedrius sip:[EMAIL PROTECTED];tag=as009e18fc. To: sip:[EMAIL PROTECTED]:5060. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 20 Jun 2006 15:15:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 576. . v=0. o=root 15254 15254 IN IP4 217.9.241.227. s=session. c=IN IP4 217.9.241.227. t=0 0. m=audio 12564 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:111 G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:110 speex/8000.
[Asterisk-Users] nortel meridian option 11c and asterisk te110p
Hi sir I am trying to interconnect meridian option 11c 2mb pri card ntbk50aa with * pri card te110p. But the problem that I am facing is that both card do not see each other the te110p card does not come out of red alarm and same is the case with meridian ntbk50aa. Hence I can not expect d-channel negociation to take place. Can u guide me some what about this as the card u used is a little different than the one I am using in m1. I have tried both t1 and e1 but same problem remains. I have tested both card so no chance of error there , cabling checked many a times no error there. When I connect some times * te110p card shows yellow/red/rec alarm. And on ntbk50aa card some times red light turn off and yellow comes on and after some times the yellow light turn off and red comes on permanently. Best Regards Mohammad Zeeshan Latif Sr. WAN Engineer NETWORK DIRECTORATE 0092-51-90391020, 0092-321-5181157 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nortel meridian option 11c and asterisk te110p
You need a cross over cable if you are linking the nortel to the te110p. http://www.merit.edu/mail.archives/nanog/2005-02/msg00546.html Julian. Muhammad Zeeshan Latif wrote: Hi sir I am trying to interconnect meridian option 11c 2mb pri card ntbk50aa with * pri card te110p. But the problem that I am facing is that both card do not see each other the te110p card does not come out of red alarm and same is the case with meridian ntbk50aa. Hence I can not expect d-channel negociation to take place. Can u guide me some what about this as the card u used is a little different than the one I am using in m1. I have tried both t1 and e1 but same problem remains. I have tested both card so no chance of error there , cabling checked many a times no error there. When I connect some times * te110p card shows yellow/red/rec alarm. And on ntbk50aa card some times red light turn off and yellow comes on and after some times the yellow light turn off and red comes on permanently. Best Regards Mohammad Zeeshan Latif Sr. WAN Engineer NETWORK DIRECTORATE 0092-51-90391020, 0092-321-5181157 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bug in asterisk static realtime?
Ciao Olle, IMHO the comments should be stripped off by asterisk itself!! It should be easy to modify the script, but the problem would remain. Should it be filed as an Asterisk bug? A semicolon in realtime separates multiple values, it is *not* used as a comment. So you should fix your script. regards, /Olle I sent this info to the script's author. Thanks for your help! -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] home routers
I use an integrated DSL modem, print sharing, firewall, wifi and 2 SIP port from DrayTek. Must be a version that has the firewalling without the modem too. Quite cheap and worked very well for 2+ years. l. On Mon, 19 Jun 2006 21:37:39 +0200, Shaun [EMAIL PROTECTED] wrote: I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: voiceone?
It looks very promising. -- -- Steven http://www.glimasoutheast.org [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Neil, I have not tried it yet, but I wanted to say this to those that don't realize it: VoiceOne is GPL http://www.voiceone.it/documentation/licence/ I just thought that was interesting... it doesn't look like it from the first look. On 6/20/06, Neil Adona [EMAIL PROTECTED] wrote: Hi! anyone from here, who uses voiceone as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help from here. That's all, thank you, Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk h323
This should provide you enough information to get started. http://www.astrecipes.net/index.php?q=astrecipes/compiling+asterisk+with+oh323 of course * can operate both SIP and h323 channels, but the support for h323 (and I'd add, stability) is not the same you can expect with SIP or IAX. l. On Tue, 20 Jun 2006 14:23:05 +0200, Khaled Chehab [EMAIL PROTECTED] wrote: Hi Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works . Please if u know send me how to active h323 protocol or the conversion protocol -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phones probem
20 jun 2006 kl. 14.28 skrev Mindaugas Kuprys: Video started to work. Now intresting thing is that video size is half reduced than calling directly from phone to phone. Phones: Tatung tia-8800. I have attached sip messages. that else might be important..? one of phones is behind nat. The reason for the half size is that we do not support the fmtp: header yet. If you want to join in the work with enhancing the video support we have a developer's list at asterisk-video, see http://lists.digium.com Thanks for the debug, it helps. In svn trunk, we have improved video support a bit, so that Asterisk won't offer another video codec on the outbound call like in this dialog, considering we don't transcode video. /Olle Btw: The following packet indicates to me that the SIP firmware in this phone is rather buggy: U 217.9.241.234:5060 - 217.9.241.227:5060 SIP/2.0 200 OK. From: Giedriussip:[EMAIL PROTECTED];tag=as009e18fc. To: sip:[EMAIL PROTECTED]:5060;tag=0-13c4-7fa-1f0bc6-74fe10e7. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Allow: INVITE,ACK,CANCEL,BYE,OPTIONS. User-Agent: TTIC Video phone. Via: SIP/2.0/UDP 217.9.241.227:5060;rport=5060;branch=z9hG4bK198aadad. Contact: sip:[EMAIL PROTECTED]:5060. Content-Type: application/SDP. Content-Length: 723. . v=0. o=Server 2890844527 2890844527 IN IP4 217.9.241.234. s=session. c=IN IP4 217.9.241.234. t=0 0. m=audio 6070 RTP/AVP 0 4 97 8 18 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:97 telephone-event/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:0 PCMU/8000. m=video 6072 RTP/AVP 34. b=AS:768. a=fmtp:34 CIF=2 QCIF=1 MaxBR=7680. a=sendrecv. a=rtpmap:34 H263/9. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integrating H.323 gateways with Asterisk?
all,
How
amenable is Asterisk to a setup that looks something like
this?
{
SIP-only VoIP hardphones } === { Asterisk } === { Cisco H.323
gateway } === { trunks to PSTN }
I've
heard Asterisk didn't play too well with H.323, but I wanted to get some more
details on that. I only recently completed my first Asterisk testbed, using four
softphones and an Asterisk box, so I'm still fairly new to
this.
thanks,
JJ
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon Paris Starts Wednesday
Just a quick reminder that AstriCon Paris starts on Wednesday morning at the Palais des Congres de Paris. The advanced team is already there and getting things ready to go. Things are wrapping up at AstriCon Berlin right now. It's been a blast. Yesterday's tutorials went well: many people learned many new things about Asterisk. Last night we had a party at the C-Base space station (C-Base is a computer and sci-fi club). Many thanks for a great time! Today we had long list of innovative and informative presentations. Thank you to the Asterisk users of Berlin for a great time. If you're in the Paris metropolitan area, please join us for a great show. We now have one-day tickets on sale for those who cannot make it for both. Thanks, Steve -- Steven Sokol CEO Sokol Associates, LLC Asterisk Training: http://www.sokol-associates.com/ AstriCon 2006: http://www.astricon.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using VoIP WiFi phones?
If anyone out there using VoIP WiFi phones? If so, which ones and what do you think about it? Thanks, W ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call rejected tone within dialplan
Hi all, I am attempting to work through some oddities with PRI signalling to neaten a few applications up and am having trouble sending a cause code 1 (unallocated) signal from within a dial plan. If I make it so that the dialled number does not match an entry in the plan I get the correct out of band continuous tone but calling SET(PRI_CAUSE=1) followed by hangup presents an engaged tone. I would have expected both to act in the same way ? Anyone got any ideas ? Tristan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integrating H.323 gateways with Asterisk?
On Tuesday 20 June 2006 15:21, J.J. Feminella wrote:
all,
How amenable is Asterisk to a setup that looks something like
this?
{ SIP-only VoIP hardphones } === { Asterisk } === { Cisco
H.323 gateway } === { trunks to PSTN }
I'm looking toward a similar setup - so far, my problems are
explained in an earlier thread sent today. There seems to be a
problem with negotiating the right codec.
Mark.
pgpLaIlukCvyS.pgp
Description: PGP signature
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX FXS.. Any experience with...
http://www.x100p.com/products_2.htm Anyone ever use this box? Hows it compare with the Iaxy? Id like to buy one or the other.. The Iaxy is appealing because to me, it seems less no name, but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? Thats appealing to the dynamic DNS guy in me! J Any experience? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: Jun 19 18:17:46 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... Jun 19 18:17:47 NOTICE[2186] chan_zap.c: Got event 2 (Ring/Answered)... Jun 19 18:17:51 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... TWO: Are there any console messages? Can you dial into the system and get internal extensions? Maybe you could try a testing dialplan like this: exten = s,1,Answer exten = s,2,Waitexten(10) exten = 100,Dial(Zap/2/014XX) Then call in and after you're connected, dial 100 to see if it will dial out on ZAP/2 When I try this, /var/log/asterisk/messages says: Jun 19 18:12:38 NOTICE[1660] pbx.c: Cannot find extension '100' in context '(null)' Jun 19 18:12:38 WARNING[1660] pbx_config.c: Invalid priority/label 'Dial' at line 172 I just realized that I blindly typed the above, without realizing that the second parameter is missing. Regardless, since even the first test doesn't work... Just in case, I'd like to repeat that I don't want Asterisk to answer the call: I just want it to use the second FXO to ring another phone, at a remote location. For reference, I went back to the original configuration that I used, but it picks up the line and remains silent (static noises): --- extensions.conf -- [cherbourg] exten = s,1,Dial(Zap/2/0145815059) --- zaptel.conf --- fxsks=1,2 loadzone=fr defaultzone=fr zapata.conf --- [channels] ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=1 ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=2 and just in case you're wondering if the FXO cards are correctly loaded... - dmesg - Jun 19 18:12:31 localhost syslogd 1.4.1: restart. Jun 19 18:12:31 localhost kernel: klogd 1.4.1, log source = /proc/kmsg started. Jun 19 18:12:31 localhost kernel: Linux version 2.6.13.4-1.x86.i686.cmov ([EMAIL PROTECTED]:1) (gcc version 3.4.4) #1 Wed Nov 23 11:31:48 EST 2005 [...] Jun 19 18:12:31 localhost kernel: Zapata Telephony Interface Registered on major 196 Jun 19 18:12:31 localhost kernel: Zaptel Version: Echo Canceller: KB1 Jun 19 18:12:31 localhost kernel: Registered Tormenta2 PCI Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt Link [LNKA] enabled at IRQ 5 Jun 19 18:12:31 localhost kernel: PCI: setting IRQ 5 as level-triggered Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt :00:08.0[A] - Link [LNKA] - GSI 5 (level, low) - IRQ 5 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt Link [LNKD] enabled at IRQ 10 Jun 19 18:12:32 localhost kernel: PCI: setting IRQ 10 as level-triggered Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt :00:09.0[A] - Link [LNKD] - GSI 10 (level, low) - IRQ 10 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: usbcore: registered new driver wcusb Jun 19 18:12:32 localhost kernel: Wildcard USB FXS Interface driver registered Jun 19 18:12:35 localhost kernel: Registered tone zone 2 (France) = Surely, I can't be the only one in this list who needs to set up Asterisk simply to ring a remote phone when a call comes in at the office. Anybody has a working configuration that I could use as a reference? Thank you :-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.1/369 - Release Date: 19/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using VoIP WiFi phones?
Warren wrote: If anyone out there using VoIP WiFi phones? If so, which ones and what do you think about it? I tried a few, but found their range and battery life to be very poor, and they were difficult to configure. I now use standard DECT phones with an ATA and they work perfectly. Two DECT handsets cost less than 50 EURO. The ATA also takes in my landline, so I only have one set of phones for both. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which is the best user GUI ?
I'm not aware of Centile using Asterisk though it could be so ...I used Centile's Callpad as an example as :1. hardware vendors (Avaya, Alcatel, ...) do not tell much about their own user GUI software2. and Centile software is often used by IP Telephony Service Providers which also use Asterisk. (For instance, Alcatel has OmniTouch Unified Communication suite which gathers My Phone, My Teamwork, My Messaging and My Assistant software. I think vendors have a hard time trying trying to sell such software : people are ready to pay for hardware but not for this kind of software) 2006/6/20, [EMAIL PROTECTED] [EMAIL PROTECTED]: Is Centile a solution built ontop of Asterisk? It looks similaraccording to their feature list. http://www.centile.com/solutions-intraswitch-platform-systemmanagement.php andhttp://www.centile.com/solutions-intraswitch-platform-advancedfeatures.php On 6/20/06, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to customise an end user application like Centiles's callpad software ( http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ? I've read www.voip-info.org User interfaces section ( http://www.voip-info.org/wiki/view/Asterisk+GUI). 23 softwares are listed. Which one is your favorite for that ? Why ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX FXS.. Any experience with...
I have a couple. The audio quality is not as good as it has a noticeable amount of hiss in the background and it also does not support message waiting. It does however support other codecs other than ulaw/alaw which is why we went for it. On Tue, 2006-06-20 at 14:51, Steve Jones wrote: http://www.x100p.com/products_2.htm Anyone ever use this box? How’s it compare with the Iaxy? I’d like to buy one or the other.. The Iaxy is appealing because to me, it seems less “no name”, but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That’s appealing to the dynamic DNS guy in me! J Any experience? __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) I have compilled for Thinstation softphone named KIAX. Switch beetwen RDP session and softphone doing like ALT-F3/ALT-F4. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On Tue, 20 Jun 2006, John Millican wrote:
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I have no
availability for high speed internet at my house to place a VoIP call over.
So, I have a standard phone plugged into the PAP2, The PAP2 plugs into the
network at my house to which the asterisk box is also connected, the
asterisk box has an FXO card that has the PSTN line plugged into it, this is
where the ZAP channel comes in. when i dial a local number asterisk simple
dials the number out the pstn line. If i dial a long distance number, the *
box dials a local phone number that I have through my VoIP provider which is
answered by an * box that I have at a different location using a line in
extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was a
little clearer this time and sorry for the confusion.
John M
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Dial command
Hello
I am trying to use this command to dial an IAX2 channel, with a supplied
context, etc:
Dial(IAX2/myiax2peer/[EMAIL PROTECTED])
This fails, with an authentication failed message while:
Dial(IAX2/myiax2peer/${EXTEN}) succeeds with out a hitch.
Why is this???
Regards
Jon
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.9.1/369 - Release Date: 19-06-2006
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Qsig
Hi All. Somebody works with asterisk linked in ISDN PRI with protocol QSIG with some PABX as Siemens, Philips, etc. The applications as pickup between asterisk and the PABX function? The names in the display and the number of the origin also? Which features that they can be used between the equipments? Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone on Thinclient?
Steve Totaro wrote: Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We have both kphone and xlite running on thinterms using LTSP nad running them as a local app, however it uses portaudio with OSS and i have noticed that different audio modules/soundcards give very different audio quality. eg CMIPCI = very good VIX82XX = very poor Bails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Add Country to CDR's
List,
Does anyone know how to add the dst Country to the CDR's via Macro (preferably).
For example, I will add a column in the cdr DB table andwhen someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column.
I have all of the International destinations in my extensions.conf like the example below:
[macro-dialout-intl]exten = s,1,SetGroup(${CALLERIDNUM})exten = s,2,CheckGroup(1)exten = s,3,absolutetimeout,${settimeout}exten = s,4,Dial(SIP/[EMAIL PROTECTED]
})exten = s,5,hangup
[intl_context]
exten = _01130.,1,Macro(dialout-intl,${EXTEN}) ;Greeceexten = _01131.,1,Macro(dialout-intl,${EXTEN}) ;Netherlandsexten = _01132.,1,Macro(dialout-intl,${EXTEN}) ;Belgium
exten = _0113271.,1,Macro(dialout-intl,${EXTEN}) ;Belgiumexten = _011331.,1,Macro(dialout-intl,${EXTEN}) ;Paris
exten = _01158212.,1,Macro(dialout-intl,${EXTEN}) ;Caracasetc...
It seems like I should be able to put the name of the country into ARG2, but I'm not sure how to write themacro to include the ARG2into the CDR.
Anyones help would be greatful.
Thanks,
bp
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk realtime and metrics
Hello guys, as you probably have already understood, I'm trying to make asterisk realtime work. Well, now it's working, but I'm not fully understanding the metrics. In voip-info.org I found that they are a sort of position inside a context (var_metric) or the index of the context (cat_metric). Am I right? Where can I obtain more info about these metrics? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Add Country to CDR's
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote: List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably). cdr(userinfo)? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing with multiple servers
Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. Any ideas on how to impliment this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller-ID Info with Voice Mail -- Can it display to the phone?
We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of office-style phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk VM system, I can get it to read back the name and number, but I'm wondering if there is a way I can get that information to display on the Cisco display as well? Off the top of my head, I can't think of any way to do this. I don't mind writing some custom XML apps either... Any one have any thoughts on this? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Add Country to CDR's
Thanks Bret, but how about an example or webpage? I'm not finding anything on google about this command for asterisk. What about AppendCDRUserField()... would this work? bp On 6/20/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote: List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably).cdr(userinfo)?--Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479 US NY +1 516 687 5200FreeWorldDialup: 635378http://www.trxtel.com the VoIP provider that pays you!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEmA/A+1olxlzQw5cRAhUkAJwLvUAipyqWAQsGcm3oBRlXeLM1KACeNyPnNziWqekYeqPA5xpUkc9jeKE==ZEqZ-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manager DBDel action
Hi, Have a look at this ticket: http://bugs.digium.com/view.php?id=6874 It contains the patch to add dbdel to your implimetation, but the command is not being added to the core of asterisk. Tim Hi list, is there a possibility to delete a key from the astdb through the manager interface? I managed to put and to get a key but I do not know how to delete an entry. The problem is that I want to use the manager interface because I can communicate remotely with my * this way. TIA, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is the current G729 compatible with Asterisk trunk?
Is the current G729 codec compatible with Asterisk trunk? /Obelix ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fun with Echo -- Follow up
I figured I'd answer my own thread and document what it took to get rid of the echo at my location. For those of you trying to get rid of echo, let me tell you, what worked for that guy, probably won't work for you. I think we've all heard that before, and it's true. Let me assure you that echo can be removed from your phone lines. At 20 hours into my 40 hours spent purging echo from my system, I didn't believe that, but its true. So, here's what it took to get it out of my system: 1. Download zaptel-trunk from SVN: As of this writing (6/20/06) using the trunk code instead of the 1.2.6 code was a major contributor in getting my echo to go away. 2. Use fxotune in zaptel-trunk: Find a silent-termination test number from the phone company and use FXOTune. I couldn't get it to dial right in order to get silence on the line. You can verify if it's working correctly by running it with an analog handset connected to your phone line. Pickup the handset and then run the command. In my case, fxotune would never clear the line, or dial the silent termination number I was giving it, not sure if this is a bug or not. What I eventually had to do was pick up the phone, dial the silent-termination number manually, run ./fxotune -i -b 4 -e 4, and quickly hangup the phone. This was the only way I got good results from the program. 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option. What eventually worked for me was the MG2 with Aggressive cancelation. 4. Along with above, you need to also try each and every combination of echocancel=xx and echotraining=xx. These setting do make a difference. I went through every possible value of each setting with each echo canceler, and kept notes along the way. 5. TIME: As I said above, I probably have 40 hours into eliminating the echo, but it is genuinely gone. Your echo problem can be solved, it will just take some time. Hope this helps someone! Thanks, Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Add Country to CDR's
On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote: Thanks Bret, but how about an example or webpage? I'm not finding anything on google about this command for asterisk. What about AppendCDRUserField()... would this work? that seems to be the same thing. the userfield lets you stick arbitrary data into your cdr records. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing calls
Hi list, I've been trying all kinds of things for hours but I keep ending up with nothing, so I was hoping to get some help. Because I could not get it to work i'v completely reset to the default configuration, except for sip.conf If I call my number I get the DEMO talking to me so I know this works.. The problem is calling out. I want to drop a call file into the spool and have the server call me and if I answer connect me to the demo (if i can get that working i probably will be able to do the rest) Can anyone tell me what i'm doing wrong, what am I missing. Regards, Marius sip.conf ### [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs allow=ilbc allow=g729 allow=gsm allow=ulaw allow=alaw allow=all ; Allow codecs in order of preference register = 31137110377:[EMAIL PROTECTED]/1000 [31137110377] type=friend context=default host=sip.budgetphone.nl fromuser=31137110377 fromdomain=sip.budgetphone.nl username=31137110377 insecure=very secret=secret qualify=no port=5060 ### This is the call-file i'm dropping: ### Channel: SIP/[EMAIL PROTECTED] Callerid: 31137110377 MaxRetries: 5 RetryTime: 300 WaitTime: 45 Context: default Extension: s Priority: 1 ### logfiles: == /var/log/asterisk/full == Jun 20 15:28:16 VERBOSE[26387]: -- Attempting call on SIP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Jun 20 15:28:16 DEBUG[26387]: Setting NAT on RTP to 0 Jun 20 15:28:16 DEBUG[26387]: Outgoing Call for 00316 Jun 20 15:28:16 DEBUG[26387]: 00316 is not a local user Jun 20 15:28:16 DEBUG[26387]: Acked pending invite 102 Jun 20 15:28:16 DEBUG[26387]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for INVITE to '"31137110377" sip:[EMAIL PROTECTED];tag=as24baf051' Jun 20 15:28:16 DEBUG[26387]: update_user_counter(00316) - decrement outUse counter Jun 20 15:28:16 DEBUG[26387]: 00316 is not a local user Jun 20 15:28:16 NOTICE[26387]: Call failed to go through, reason 8 Jun 20 15:28:16 DEBUG[26387]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for CANCEL == /var/log/asterisk/messages == Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for INVITE to '"31137110377" sip:[EMAIL PROTECTED];tag=as24baf051' Jun 20 15:28:16 NOTICE[26387]: Call failed to go through, reason 8 Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for CANCEL == /var/log/asterisk/full == Jun 20 15:28:31 DEBUG[26387]: Auto destroying call '[EMAIL PROTECTED]' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using VoIP WiFi phones?
On 09:41, Tue 20 Jun 06, Warren wrote: If anyone out there using VoIP WiFi phones? If so, which ones and what do you think about it? We dont use them because battery time is bad bad bad. We use dect phones with an ATA and the tiptel/kirk dect set. They work perfectly. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: fail to make call
Hi ... In my configuration below, I use realtime architecture in our system. I have one device attached to each asterisk server. There is no record when I issue sip show users or sip show registry in CLI. I wonder how can I know who is registered in asterisk. What command is it? On 6/20/06, unplug [EMAIL PROTECTED] wrote: Hi I have the following configuration | UA1 --|-- asterisk1 ---+ UA2 --|-- asterisk2 ---+ DB UA3 --|-- asterisk3 ---+ UA4 --|-- asterisk4 ---+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register infomration. UA1 can make call to UA2,UA3 and UA4. UA2 can make call to UA1, UA3 but not UA4. UA3 can make call to UA4 but not UA1, UA2 UA4 failed to make call to all UA. From the CLI and log below, asterisk shows it can't create channel. As I expect, all UA should able to find each other. However, some of them are failed to find others. I have no idea why they can't find each other well. Is it the configuration problem? Anyone can help? -- Executing Dial(SIP/871966629896-5373, SIP/871966760539|15) Jun 20 17:42:16 NOTICE[23355]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/871966629896-5373, ) NOTICE[28269] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo -- Follow up
I eliminated my echo almost instantly by purchasing an echo canceling card :) I had about 30 minutes into to get the card installed and asterisk up and running. On 6/20/06, Brian Swan [EMAIL PROTECTED] wrote: I figured I'd answer my own thread and document what it took to get rid of the echo at my location. For those of you trying to get rid of echo, let me tell you, what worked for that guy, probably won't work for you. I think we've all heard that before, and it's true. Let me assure you that echo can be removed from your phone lines. At 20 hours into my 40 hours spent purging echo from my system, I didn't believe that, but its true. So, here's what it took to get it out of my system: 1. Download zaptel-trunk from SVN: As of this writing (6/20/06) using the trunk code instead of the 1.2.6 code was a major contributor in getting my echo to go away. 2. Use fxotune in zaptel-trunk: Find a silent-termination test number from the phone company and use FXOTune. I couldn't get it to dial right in order to get silence on the line. You can verify if it's working correctly by running it with an analog handset connected to your phone line. Pickup the handset and then run the command. In my case, fxotune would never clear the line, or dial the silent termination number I was giving it, not sure if this is a bug or not. What I eventually had to do was pick up the phone, dial the silent-termination number manually, run ./fxotune -i -b 4 -e 4, and quickly hangup the phone. This was the only way I got good results from the program. 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option. What eventually worked for me was the MG2 with Aggressive cancelation. 4. Along with above, you need to also try each and every combination of echocancel=xx and echotraining=xx. These setting do make a difference. I went through every possible value of each setting with each echo canceler, and kept notes along the way. 5. TIME: As I said above, I probably have 40 hours into eliminating the echo, but it is genuinely gone. Your echo problem can be solved, it will just take some time. Hope this helps someone! Thanks, Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Vincent Delporte wrote: Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: You have a problem unrelated to what you are trying to do. Fix the problem with dialing out of Zap/2 first. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo -- Follow up
On Tuesday 20 June 2006 11:30, Brian Swan wrote: 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option. What eventually worked for me was the MG2 with Aggressive cancelation. I hate to tell you this, but if you have turned on the aggressive suppressor you aren't cancelling echo. You have turned your phone into a half-duplex communication medium. With the aggressive suppressor enabled, when zaptel detects you talking, it MUTES the received audio. Try it -- call up a friend and ask him to burp the alphabet. While he's doing that, talk to him. You will stop hearing him whenever you talk. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Correct me if I'm wrong but I think you would want to use the transfer command instead of dial to get it to call out to a remote office. -John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User Loses Ability to Make Outgoing Calls
We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no interference from us. She's using a Linksys PAP2-NA like the rest of us, and we've tried changing her adaptor, but the problem persists. The only thing I can think of is that it has to do with the way her internet connection is set up. She is using a D-Link wireless router (but of course the adaptor is not through the wireless part); 802.11g/2.4GHz Does anyone know of anything that could be triggering this odd behavior or have more detailed questions I can ask her to help pinpoint the problem? Any assistance is much appreciated. Leah Newmark Capalon VoIP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID Info with Voice Mail -- Can it display to the phone?
Message: 21Date: Tue, 20 Jun 2006 10:12:38 -0500From: Brian Swan [EMAIL PROTECTED]Subject: [Asterisk-Users] Caller-ID Info with Voice Mail -- Can itdisplay to the phone?To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed We recently switched my wife's business over to an Asterisk setupusing Cisco IP phones (7940s and 7960s) with chan_sccp.They didn'tuse any kind of office-style phone system before, they had one phone in the office with a built in answering machine that woulddisplay the Caller ID of the person who left the message whileplaying the message.I know in the Asterisk VM system, I can get itto read back the name and number, but I'm wondering if there is a way I can get that information to display on the Cisco display as well?Off the top of my head, I can't think of any way to do this.I don'tmind writing some custom XML apps either...Any one have any thoughts on this? Thanks!BrianBrian-I've been working on this for some time- and it is possible, although has a few pitfalls. I'm not done with the Asterisk VM object to support it yet, but I have developed a Services menu item for 79XX series phone (using CMXML3- it won't work for SIP loads yet- although I havent tested it with an 8.0 series load) that will show you the voicemail by caller id, allow you to select (cherry-pick, as it were), then playback, delete, mark read/unread, and return the call. The system is backended by a php class I wrote to work with Cisco Unity- but I'm just about to start work on the class for Asterisk VM. (at my current pace, expect it in a few months). The system is developed privately- not GPL, not for sale- done for a client- but I am free to discuss the methods and madness within it, or develop a parallel system and contribute bits back to the OSS community- it's just tied in larger proprietary system which would be difficult to chop out without some time I don't have in my schedule right now. If the Asterisk class development goes well, I may do so- but it won't happen for a while. I'd be happy to discuss the implementation details with you if you'd like.-Paul DavidsonPlanCommunications, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ECHO Tutorial
On Mon, 2006-06-19 at 18:45 -0400, Gary Reuter wrote: On 6/19/06, Daniel Salama [EMAIL PROTECTED] wrote: Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? This paper by Cisco is a great start: Echo Analysis for Voice over IP http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml (it's the first result I get when I google for echo in voip) There was also a good article in LJ late last year: http://www.linuxjournal.com/article/8424 -Seth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone on Thinclient?
Vitaly, That is good news, but I'm afraid that switching between screens will be a bit too much for my end users to handle. On 6/20/06, bails [EMAIL PROTECTED] wrote: Steve Totaro wrote: Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We have both kphone and xlite running on thinterms using LTSP nad running them as a local app, however it uses portaudio with OSS and i have noticed that different audio modules/soundcards give very different audio quality. eg CMIPCI = very good VIX82XX = very poor Bails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] teste E1 card
Hi,Can I, just for test, use a crossover cable linking 2 channels of my E1 card (TE406P) and dial from one channel to another?Is there any different way to do this?-- Ralph Liebessohn ICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using VoIP WiFi phones?
On 6/20/06, Warren [EMAIL PROTECTED] wrote: If anyone out there using VoIP WiFi phones? If so, which ones and what do you think about it? As others have said, they are all horrible. If you /must/ have one, the Hitachi WIP3000 or WIP5000 both do the job. AFAIK these are the only phones with a stable WPA implementation, and although they are sensitive to signal before making a call, they are very good at holding on to keep a call going under fairly harsh circumstances. I still would not put my name to a system with a WiFi phone in it if I could help it though. Too much trouble :( Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-backports.org
hi all I just setup a new site, perhaps soon a wiki, to collect what's out there of useful backports from Trunk/1.4 beta back to 1.2. Take a look at http://http://www.asterisk-backports.org/ and judge for yourself ;) roy -- Roy Sigurd Karlsbakk [EMAIL PROTECTED] (+47) 98013356 --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 doesn't register after reboot
Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call limit function on sip channel to external pop
At 12:20 AM 6/20/2006, you wrote: Anyone already had such an issue, or anyone knowing the best config for limiting outgoing sip channels to external sip providers? It's kind of urgent... I did that using groups in the dialplan. There's an example under group at the wiki I did that might help. Ira http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sangoma unicall m2rfc
Steve. Im also getting a lot of these: Jun 20 10:34:58] WARNING[16786]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing [Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected [Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2930 handle_uc_event: CRN 32818 - far disconnected cause=Switching equipment congestion [42] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwood |Sent: Monday, June 19, 2006 7:15 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] sangoma unicall m2rfc | |Anton Krall wrote: | |Uys, Steve Underwood | |I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for |R2MFC, I get the far and local end unblocked but as soon as I try to |make a call I get dialing and then protocol failure.. | |Do you guys know if there are any issues with sangoma and unicall? |Anybody has an a101 card working with unicall and r2mfc? | |Are you out there Steve? :) | | | |Lots of people are using Sangoma cards successfully with Unicall. | |Regards, |Steve |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
http://www.voip-info.org/wiki/index.php?page=Australia%20Asterisk% 20Details Stumbled across this Reverse On Idle Condition (ROIC) 'feature' that sounds very promising. Will get it enabled later today and give it a go. On Tue, 2006-06-20 at 23:35 +1000, Carey O'Shea wrote: Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
Re: [Asterisk-Users] SIP Softphone on Thinclient?
Steve Totaro a écrit : Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? May I advertise MozIAX (moziax.mozdev.org) ? It is well suited to thin client environment, because the user interface (Firefox extension) and the engine (iax and sound management) communicate through network, so you can run the UI on the server, and the engine on the thin client, and you don't need to run a network sound system on the thin client. I think it gives better sound quality. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo -- Follow up
On 6/20/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 20 June 2006 11:30, Brian Swan wrote: 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option.What eventually worked for me was the MG2 with Aggressive cancelation.I hate to tell you this, but if you have turned on the aggressive suppressoryou aren't cancelling echo.You have turned your phone into a half-duplexcommunication medium.With the aggressive suppressor enabled, when zaptel detects you talking, it MUTES the received audio.Try it -- call up a friend and ask him to burp the alphabet.While he's doingthat, talk to him.You will stop hearing him whenever you talk.-A. OH God, 40 hours lost !-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk h323
You can do this by installing a h323 module. Conversion works simetimes good, sometimes not good. H323 behaviour on asterosk with my experience with kind of unpredictable. 2006/6/20, Khaled Chehab [EMAIL PROTECTED]: Hi Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works . Please if u know send me how to active h323 protocol or the conversion protocol Regards *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Hakem Voip ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ECHO Tutorial
In the context of Asterisk and TDM cards, I think this article is pretty good. Very light on the technical but David points out some of the unique challenges. http://www.linuxjournal.com/article/8424 -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ECHO Tutorial Daniel Salama wrote: Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? Why is this affecting the VoIP world so much and not the regular PSTN analog world? What does the PSTN industry have that they can handle such high volume of calls and there is no echo problem? Search the list archives, there is more then enough information there. http://lists.digium.com/pipermail/asterisk-users/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Provisional problem with SIP channel
Hi, I'm using the Perl AGI interface for a prepaid card platform. And sometimes (almost twice an hour), asterisk doesn't detect a call has been hung up. The call is so hung up when the time limit for the call is reached (the corresponding prepaid card is then emptied ...). I've tried to look in the asterisk log files to find anything suspect with these calls, and I've found a debug message which looks to be my problem : chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) Each time I've got that, the call is not normally hung up. Hopefully that's not a bug in asterisk ! Is there anyone who could help me ? It's hard to extract logs as it's hard to identify these few calls (that's not a constant problem). Here are some logs I've extracted : Jun 20 15:59:07 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as03ec066e Jun 20 15:59:07 DEBUG[25077] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 20 15:59:07 DEBUG[25077] chan_sip.c: SIP response 100 to standard invite ... Jun 20 15:59:11 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as3eafcd68 Jun 20 15:59:11 DEBUG[25077] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 20 15:59:11 DEBUG[25077] chan_sip.c: SIP response 100 to standard invite Jun 20 15:59:17 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as3eafcd68 Jun 20 15:59:17 DEBUG[25077] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 20 15:59:17 DEBUG[25077] chan_sip.c: SIP response 183 to standard invite ... Jun 20 15:59:19 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag 169705f8 Our tag: as3eafcd68 Jun 20 15:59:19 DEBUG[25077] chan_sip.c: Acked pending invite 102 Jun 20 15:59:19 DEBUG[25077] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 20 15:59:19 DEBUG[25077] chan_sip.c: SIP response 200 to standard invite Jun 20 15:59:19 DEBUG[25077] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED];transport=UDP ... Jun 20 15:59:22 DEBUG[25077] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 169705f8 Our tag: as3eafcd68 Jun 20 15:59:22 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as03ec066e Jun 20 15:59:22 DEBUG[25077] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 20 15:59:22 DEBUG[25077] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 169705f8 Our tag: as3eafcd68 Jun 20 15:59:22 DEBUG[25077] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as03ec066e Jun 20 15:59:22 DEBUG[25077] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jun 20 15:59:22 DEBUG[25077] chan_sip.c: Updating call counter for outgoing call Thanks, Benoit -- Benoit Merouze _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._. Groupe IPercom - The VoIP Enabling Company - http://www.ipercom.com Ingénieur RD - courriel : [EMAIL PROTECTED] Network Software Developer - mailto: [EMAIL PROTECTED] Tél. / Phone : +33 1 7269 9611 ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Siège Social 43, rue Fessart 92100 Boulogne Billancourt RCS NANTERRE B 440 345 528 - Capital social: 100 000 € CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES SONT CONFIDENTIELS COUVERTS PAR LE SECRET PROFESSIONNEL THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Only two things are infinite, the universe and human stupidity, and I'm not sure about the former. Albert Einstein ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P bad echo problem, tried lots of things
I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time restarting Asterisk, then opening the Zap channel, and then speaking...only to hear my self played back almost instantly. None of these options changed the echo for me, it always sounded the same -- except for the AGGRESSIVE_SUPPRESSOR option, in which every time I spoke it made the other end a very low volume, so much that I couldn't hear the other end (ie: not useful). I don't have this problem with pure IP calls, it's only with my TDM400P and FXO that I have this echo problem. This means my headset and IP phones are fine (of course). So, what else can I try? :-) Any ideas why this is so consistent and persistent? Maybe it's something to do with my phone cable or something of that nature (hmm?)? Any input appreciated. Thanks, Carey O'Shea. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2.9.1 crashed today
I upgraded to 1.2.9.1 today. It was working fine until after lunch. After running since 8am it stopped around 1pm. People could still call in on our PRI via Zap. But, you couldn't use the dialplan (would just sit there)... the queue went to dead air.. and 'show agents' 'show queues' 'zap show channels' did nothing.. just returned back to the asterisk prompt with no information. Calls there were up (queue calls) continued to work. Any thoughts on what might have happened? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 doesn't register after reboot
Mimmus wrote: Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I think that was fixed in 6.2.1. See http://www.snom.com/wiki/index.php/Beta_Firmware and http://www.voip-info.org/wiki/view/snom+360 - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX FXS.. Any experience with...
On Jun 20, 2006, at 6:51 AM, Steve Jones wrote: x-tad-smallerhttp://www.x100p.com/products_2.htm/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAnyone ever use this box? How’s it compare with the Iaxy? I’d like to buy one or the other.. The Iaxy is appealing because to me, it seems less “no name”, but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That’s appealing to the dynamic DNS guy in me! /x-tad-smallerx-tad-smallerJ /x-tad-smallerx-tad-smallerAny experience? /x-tad-smaller I hadn't seen that one! I have the AG168V based ATA which with recent firmware update (1.51) is working very well with IAX2, although there is still an occasional crack sound on calls. I am very happy with the AG168v, which also a generic cheapo unit with a universal power adapter (I wonder if that's where the crack sound comes from), and with recent firmware audio quality is MUCH improved for IAX2 (I was using SIP for a while). I wonder if the unit you are linking to is also based on PA168 chipset? seems kind of likely to me. I also wonder about setting up the switch for ethernet purposes (ie does it use DHCP? can I set it to be static? can it port forward to the connected PC?,etc). Thanks for the link, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX FXS.. Any experience with...
On Jun 20, 2006, at 7:09 AM, Gareth Blades wrote: I have a couple. The audio quality is not as good as it has a noticeable amount of hiss in the background and it also does not support message waiting. I have looked at the docs and this appears to be identical to the AG168V with regards to the setup screens. Have you tried out the latest firmware from aredfox.com? The 1.51 firmware radically improved my voice quality for the device. Strangely it changed the ringtone to a non US sounding one, which is kind of slick for my installation, as it now acts as a distinctive ring feature (ie regular sounding ring PSTN incoming, strange short ring Voip call). Here is a link (third from bottom is IAX firmware) http://aredfox.com/edownloads.htm It does however support other codecs other than ulaw/alaw which is why we went for it. Another feature that I like about this device is that it can be flashed to an SIP ATA and back to IAX without losing all of your settings... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using VoIP WiFi phones?
I have been pretty happy with my cisco 7920, but it has been by the wayside for six months or more now due to the wimpy battery life. I recommend a standard cordless phone (yes, even 2.4ghz) and ATA to beat the wifi voip phones I've tried :( Warren wrote: If anyone out there using VoIP WiFi phones? If so, which ones and what do you think about it? Thanks, W ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 doesn't register after reboot
Hi Domenico, Try Ver. 6.2.1. This problem is fixed in it. http://www.snom.com/wiki/index.php/Beta_Firmware#Release_6.2.1 Regards, Usman Tahir snom technology AG -- Message: 17 Date: Tue, 20 Jun 2006 18:18:43 +0200 From: Mimmus [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 360 doesn't register after reboot To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo -- Follow up
OH God, 40 hours lost ! Yup.. at 20$/hour that's 800$ that could have been put into a better piece of hardware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo -- Follow up
what's that ? where did u purchase is ? right now i'm having echo problems between two asterisk servers dealing with iax with ulaw codec, one in italy and the second in thailand in your opinion, it is possible that an echo issue is derived by low bandwidth ? i thought this will end having delay/choppy behaviour, not echo ! do you think that could be an hardware problem related with my ip phones ? i'm using taiwanese phones, the tip-100, from that list: http://www.ttic.com.tw/program/en_product_catalog.asp?product_series_h=internetproduct_layout=list tnx to all feedback ! .mike On Tue, 2006-06-20 at 11:43 -0400, Matt wrote: I eliminated my echo almost instantly by purchasing an echo canceling card :) I had about 30 minutes into to get the card installed and asterisk up and running. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User Loses Ability to Make Outgoing Calls
On Jun 20, 2006, at 8:53 AM, Leah Newmark wrote: We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no interference from us. She's using a Linksys PAP2-NA like the rest of us, and we've tried changing her adaptor, but the problem persists. The only thing I can think of is that it has to do with the way her internet connection is set up. She is using a D-Link wireless router (but of course the adaptor is not through the wireless part); 802.11g/2.4GHz Does anyone know of anything that could be triggering this odd behavior or have more detailed questions I can ask her to help pinpoint the problem? Probably a crappy internet connection and the ATA is becoming unregistered from the server. Got your post twice by the way? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
