[Asterisk-Users] Re: zaptel make problems
[EMAIL PROTECTED] is believed to have said: The error is the same, afaik. What I can't understand is why the make is entering in the directory '/ usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert, but I would expect it to go fiddle with a '586' directory. Just a guess, your simlink is pointing to the incorrect linux source directory. Go into /usr/src/linux and do a ls -l, see where the linux simlink is pointing to. If it's incorrect, then do a rm linux and delete it. Recreate with a ln -s /usr/src/yourlinuxversionhere Doug Doug, thanks for the recipe! The weird thing is that a similar setup of SUSE 9.3, albeit not updated, did not show this quirk. Now, it should however be easy even for me to fix things. Thanks again Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaptel make problems
[EMAIL PROTECTED] is believed to have said: and watch linus himself rant about how this is incorrect to do (yet all the distros do it) :P Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXY Voicemailmain problem
On Thu, 2005-21-07 at 23:08 -0500, Steve Maroney wrote: I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03 WARNING[5111]: res_adsi.c:292 __adsi_transmit_messages: Unexpected response to ack: (retry 2) and then after a few tones I see: Jul 21 23:06:04 WARNING[5111]: res_adsi.c:296 __adsi_transmit_messages: Maximum ADSI Retries (3) exceeded and then the app conttinues : -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/9' (language 'en') So Im guessing its something to do with ADSI. So far, I only have this problem when checking voicemail, not for outgoing calls to another voip--pstn gateway. Thank you, Steve Maroney Indeed, that's the Comedian Mail ADSI scripting being sent to the device as in-band FSK tones, just like CallerID. I know in zapata.conf, you can specify adsi=no, but I don't think you can do it for iax. What's more, IIRC, the ADSI scripting is hard-coded into app_voicemail.c and therefore wouldn't be affected by the previously mentioned ADSI setting. So it looks like you're stuck with it, unless either I'm wrong and you can turn off app_voicemail's ADSI functions, or you go ahead and patch up an ADSI-free version. -Bryce [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues and roundrobin/rrmemory
I have a queue setup using Asterisk CVS and roundrobin, however calls seem to be distributed in the same way as rrmemory (round robin with memory), ie, it is alternating between the two people in the queue rather than always calling the first available person in the queue first. I am using agents with agentcallbacklogin and addqueuemember to dynamically add the agent to the queue. asterisk version: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 2005-06-07 07:34:45 Does anyone use agents + agentcallbacklogin and use roundrobin queues with a recent CVS and have it working (or have the same problem ??) Thanks, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP messengers video phones
Juraj Bednar wrote: Hello, There's some work on creating a multiprotocol solution for instant messaging within Asterisk, but it will not be in the coming v1.2. is the work somewhere as a patch to be tried or in some other form, even if it's not coming to 1.2? No, there needs to be some serious additions to the core, so I don't think there will be patches for previous versions of Asterisk. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No D-channel available
Hi, I installed a Quadbri card, configured with 2 ports connected to a Hipath pbx an 2 ports connected to telco. I can make and receive call but I receive every 5 seconds on asterisk cli the message: No D-channel available Using Primary on channel aniway 12 Primary D-channel on span 4 up My zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 context=isdn-incoming group = 1 immediate=no signalling = bri_cpe channel = 1-2 channel = 10-11 context=pbx-incoming group = 2 immediate=no signalling = bri_net channel = 4-5 channel = 7-8 zaptel.conf loadzone=it defaultzone=it # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xorcom Rapid 1.1
Hi all Xorcom Rapid 1.1 is here. * Asterisk 1.0.9 * Flash Operator Panel * improved Zaptel hardware detection: should hopefully detect E1, T1 ZapHFC and qozap. No more channel numbers guessing in zapata/zaptel.conf * and much of the extra software available for it You can get the full details at http://www.xorcom.com/rapid so I'll just highlight the points that I believe are more relevant to the people here rather than a standard press-release. Warning: long post. Xorcom Rapid is based on the current Debian Stable. This is not just Asterisk built from source on a certain system: we use native distribution packages. You can install just the parts you like. E.g: spandsp and the h323 channel (with their extra dependencies) are optional components. It is also possible to upgrade packages or the whole system. It is a binary distribution. Some people really don't like that idea. They think that if you didn't built it from upstream source it's not worth it. Well, if you have such an attitude then why are you running a Linux/*BSD distribution? use LinuxFromScratch to build a system like a real programmer, and come back to report how long it took you and if you eventually did get a better and more manageable result. Binary distribution is by no means locked down. You need to apply some fixes to the source? a decent packaging system provides simple ways of extracting the original source, patching it, and building the result. Build your own debs. True, you may need to set up a separate build system, but then again, the whole build tool-chain is not needed for a PBX to run. We tried to separate the configuration to smaller files. This should make it safer to use newer configuration that fixes and enhances the default, and yet maintain your local changes. We do want to make it easier for you to upgrade your system, so you won't be stuck with an old, broken Asterisk that happened to work and you don't touch it. That said, we do realise that the voodoo factor is still considerably large. We can't and won't try forcing upgrades on anybody's precious PBX system. This version is based on 1.0 . However it seems that the CVS head is really not that far from becoming 1.2 . The next version of Rapid will be based on it. Debian is also supposed to start working with Asterisk 1.1 packages in the Experimental branch. In the near future we will probably continue backporting required packages from Unstable when necessary and maintain compatibility with Stable. Vim is included, along with syntax highlighting for asterisk configurations. vim is not the default vi (nvi is much smaller, you know) but if you edit many files, you'd probably want to install it. I am looking for improvements: e.g: when editing Apache's httpd.conf files or CSS files, the syntax highlighting is very good at spotting syntax errors. I have already added something simple in that direction (a line that begins with '#' and is not an 'include' will be coloured as an error), but I'd like to see more. Also included in this release is a web-based configuration interface called DeStar . I'm interested to expose it to a larger crowd, so have a go with it. I've included some scriptary to play convert MP3s (off line) to phone quality WAVs, and to play the WAVs with sox for the music-on-hold . I would appreciate input on what you'd expect there. e.j: add some randomisation to the wav-player scriptlet? The detection of Zaptel PRI and BRI cards should detect channel numbers correctly. But the span parameters and such are generally my simple attempt to give sane defaults. If it doesn't work in your case, please let me know. As a general note, if a simple shell script can detect channel numbers so easily, why can't chan_zap do all the work by itself? And another small thing to simplify the initial testing: iaxcomm.exe is included on the CD. For windows people it should run off the CD. One less thing to download. We have also set up a mailing list for Rapid, so feel free to subscribe there and post questions to a smaller, more focused crowd: http://xorcom.com/mailman/listinfo/users_xorcom.com -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | http://www.xorcom.com/| friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel make problems
On Fri, 2005-07-22 at 07:59 +0200, Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: and watch linus himself rant about how this is incorrect to do (yet all the distros do it) :P Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Who said it's an error, Linus just does not like it and thinks says it's incorrect, it causes no errors, and when you have multiple kernel sources on the same machine it makes life much easier. I would agree that going through multiple symlinks is bad practice, this could also be Linus' argument, or maybe it's multiple times through the same symlink in the case of a kernel compile. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
It's not a spam. They are not yokels. Don't know about you Gizmo is basically a different front end offered by the Sipphone.com people, to offer an alternative to Skype which is not a closed jail (interoperates with all SIP devices, asterisk, etc.). I think they sent the mail to all registered sipphone.com users. On Thu, 21 Jul 2005, Jay Milk wrote: Got an email this morning with the subject Welcome to Gizmo Project. I didn't sign up with those yokels. Anyone else got spammed by them? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel make problems (long)
On Thu, Jul 21, 2005 at 04:52:43AM -0700, trixter http://www.0xdecafbad.com wrote: On Thu, 2005-07-21 at 07:36 -0400, Doug Lytle wrote: Just a guess, your simlink is pointing to the incorrect linux source directory. Go into /usr/src/linux and do a ls -l, see where the linux simlink is pointing to. If it's incorrect, then do a rm linux and delete it. Recreate with a ln -s /usr/src/yourlinuxversionhere ln -s localname /usr/src/linux Avoid absolute symlinks if you don't have to. and watch linus himself rant about how this is incorrect to do (yet all the distros do it) :P If all the distros do it, then maybe it's not such a bad idea. Reminds me of Linus' rant about /usr/include/linux which wasn't exactly valid either. IMHO, the aptel build system is wrong here. It makes a number of (different) assumptions as to where the source resides . Sometimes it is /usr/src/linux , sometimes its the target of /lib/modules/`uname -r`/build I've fixed those in my local version and I build it vs a number of different kernel trees with no problem at all. Debian, for once, does not symlink /usr/src/linux for you. There should really be no need. It also provides nice kernel-headers packages, that should include everything you need from the configured kernel tree to build modules against. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, Jul 21, 2005 at 09:45:02PM +0100, Angus Comber wrote: I am now getting this make error: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory Remove the line that includes asterisk.h . Doesn't help anybody. This is basically the patch I needed to apply to asterisk-addons to make it build with the debian package asterisk-devel . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: zaptel make problems
On a different note using Fedora Core 3 I get CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_chan_write': /usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `ioctl_load_zone': /usr/src/zaptel/zaptel.c:2392: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `zt_common_ioctl': /usr/src/zaptel/zaptel.c:2744: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2804: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2807: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2889: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2919: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `zt_chanandpseudo_ioctl': /usr/src/zaptel/zaptel.c:3641: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3651: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3654: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3713: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3717: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: At top level: /usr/src/zaptel/zaptel.c:176: warning: 'fcstab' defined but not used When building the stable or head zaptel with kernel linux-2.6.11-1.35_FC3. The module compiles but it never used to give this message on FC2. Anyone got any ideas? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 22 July 2005 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: zaptel make problems On Fri, 2005-07-22 at 07:59 +0200, Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: and watch linus himself rant about how this is incorrect to do (yet all the distros do it) :P Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Who said it's an error, Linus just does not like it and thinks says it's incorrect, it causes no errors, and when you have multiple kernel sources on the same machine it makes life much easier. I would agree that going through multiple symlinks is bad practice, this could also be Linus' argument, or maybe it's multiple times through the same symlink in the case of a kernel compile. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: zaptel make problems
On Fri, 2005-07-22 at 08:31 +0100, Lee Archer wrote: On a different note using Fedora Core 3 I get When building the stable or head zaptel with kernel linux-2.6.11-1.35_FC3. The module compiles but it never used to give this message on FC2. Anyone got any ideas? compiler upgrade between FC2 and FC3? Latest version of gcc et al is more picky than before, Kevin and co. are working on it. (Well I hope so ;-) ) Just an aside, the kernel also gets some of these types of messages. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Zaptel FXO..
Hi all, i've installed AMP and Asterisk following the INSTALL file and i have a problem with the TDM04B with 4 FXO: [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) [EMAIL PROTECTED] ~]# and this is my lsmod: [EMAIL PROTECTED] ~]# lsmod Module Size Used by wcfxo 16928 0 zaptel195972 1 wcfxo parport_pc 27905 1 lp 15405 0 parport37641 2 parport_pc,lp autofs422085 0 i2c_dev14273 0 i2c_core 25921 1 i2c_dev sunrpc138789 1 ipt_REJECT 10561 1 ipt_state 5825 4 ip_conntrack 45701 1 ipt_state iptable_filter 6721 1 ip_tables 21441 3 ipt_REJECT,ipt_state,iptable_filter dm_mod 58949 0 button 10449 0 battery12869 0 ac 8773 0 md5 8001 1 ipv6 238817 24 uhci_hcd 32729 0 ehci_hcd 31813 0 hw_random 9557 0 hisax 413297 0 crc_ccitt 6081 2 zaptel,hisax isdn 125473 1 hisax slhc 11201 1 isdn tg382373 0 floppy 58065 0 ext3 118729 2 jbd59481 1 ext3 raid1 19521 2 mptscsih 36605 0 mptbase47457 1 mptscsih sd_mod 20545 7 scsi_mod 116429 2 mptscsih,sd_mod [EMAIL PROTECTED] ~]# It's a RH ES 4.0..someone can help me ? Thanks ! Oz -- O-Zone ! No (C) 2005 www.zerozone.it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP2000 and Headsets, Call Center phones.
I see the GXP2000 has a headset socket. Are their any compatible headsets for it. How does the functionality change? What else would people suggest for a Call-Centre? Would like Headset, Call Details - etc... The call centre answers the phone according to which number is called.. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Zaptel FXO..
Hi, please post you zaptel.conf Giorgio Michele O-Zone Pinassi wrote: Hi all, i've installed AMP and Asterisk following the INSTALL file and i have a problem with the TDM04B with 4 FXO: [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) [EMAIL PROTECTED] ~]# and this is my lsmod: [EMAIL PROTECTED] ~]# lsmod Module Size Used by wcfxo 16928 0 zaptel195972 1 wcfxo parport_pc 27905 1 lp 15405 0 parport37641 2 parport_pc,lp autofs422085 0 i2c_dev14273 0 i2c_core 25921 1 i2c_dev sunrpc138789 1 ipt_REJECT 10561 1 ipt_state 5825 4 ip_conntrack 45701 1 ipt_state iptable_filter 6721 1 ip_tables 21441 3 ipt_REJECT,ipt_state,iptable_filter dm_mod 58949 0 button 10449 0 battery12869 0 ac 8773 0 md5 8001 1 ipv6 238817 24 uhci_hcd 32729 0 ehci_hcd 31813 0 hw_random 9557 0 hisax 413297 0 crc_ccitt 6081 2 zaptel,hisax isdn 125473 1 hisax slhc 11201 1 isdn tg382373 0 floppy 58065 0 ext3 118729 2 jbd59481 1 ext3 raid1 19521 2 mptscsih 36605 0 mptbase47457 1 mptscsih sd_mod 20545 7 scsi_mod 116429 2 mptscsih,sd_mod [EMAIL PROTECTED] ~]# It's a RH ES 4.0..someone can help me ? Thanks ! Oz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX over HTTP
You can traversal a HTTPS proxy using a plain TCP connection (without SSL). The unique requirement of some HTTPS proxys is that the target port is 443. Then if your Asterisk listen in 443 port IAX (TCP) connections, it should work. G. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Julio Arruda Enviado el: jueves, 21 de julio de 2005 23:53 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] IAX over HTTP Rob Scott wrote: For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much. I did some tests, with openvpn, for my purpose, was ok, not sure how would behave in packet loss, jitter conditions.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SATA
Title: SATA Has anyone had any problems with SATA, either on board or 3rd party setup? I've currently got a problem where an AMD non SATA FC2 system is working fine but an Intel system with a 3Ware SATA card and FC3 is radomly not syncing with the ISDN30. It allows and receives calls but at random intervals drops them. Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and IP500 / IP600 Boot RoM
Hi, if you use version 3 you cannot go back to previous version without sending your phone to Polycom. If you want we have some files but we do not guarantee the right working. Which version do you need? Giorgio Michael Felder wrote: Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] http://www.ITMedic.com.au http://www.itmedic.com.au/ Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming calls
hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg: Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??, thx in advance, CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de 1,49 euros par mois___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No dialtone - iaxy
Hi, try to add the gateway field in your config file... My conf file: ip: 192.168.100.2 netmask: 255.255.255.0 gateway: 192.168.100.3 codec: ulaw server: 192.168.100.3 user: iaxy pass: iaxy register Giorgio Bryce Chidester wrote: On Wed, 2005-07-20 at 08:26 -0400, Ousmane Doukara wrote: Hi, I am unable to get a dialtone from iaxy (the old model). When dial a mailbox, I can see the mailbox app reacting. iaxy gets registered. I can make call and the remote phone can hear me. No sound for iaxy user. ./iaxyprov 192.168.1.134 provinfo 01: 05: 11 d9 0d: 00000004 0f: 4546d2e7 10: 11 d9 06: 6d616c69 07: 636f756d62613738 0c 0000000d Provisioning is 44 bytes Total packet is 58 bytes Got response back from 192.168.1.134 --- dhcp codec:ulaw server:192.168.1.140 user:username pass:pass register -- Any idea ? Sounds like your IAXy is fried. This was/is a fairly common issue with the old model when left on long enough. Presumably, this was the reason for the redesign. Of course, this could be a completely different issue, but the symptoms match. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk captures sound device
Hello, dear Asterisk experts. When I run Asterisk (CVS HEAD version), I'm not able to play music anymore -- asterisk seems to capture sound device. Is it not a bug, but a feature? That's unlike stable (1.0.7 and 1.0.9) versions, when I can, say, run an IP telephone on the *same* machine and listen what Asterisk' autoattendant says. Now I can't do that, I need Asterisk and client running on the separate machines. Thank you in advance for clarifying the problem. -- Best regards, Timur Elzhov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Zaptel FXO..
On Friday 22 July 2005 11:37, Giorgio Incantalupo wrote: Hi, please post you zaptel.conf Giorgio Thanks i've solved: i'm using a wrong modules combination :-D Thanks however ! Oz -- O-Zone ! No (C) 2005 www.zerozone.it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
How strange - that worked! I wonder why that was put there? Angus - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 22, 2005 8:29 AM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons On Thu, Jul 21, 2005 at 09:45:02PM +0100, Angus Comber wrote: I am now getting this make error: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory Remove the line that includes asterisk.h . Doesn't help anybody. This is basically the patch I needed to apply to asterisk-addons to make it build with the debian package asterisk-devel . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk operator functions
Hey! My asterisk is working properly so far with all automatic functions. Now I want to direct incoming calls to operator, i mean some person who answers to the incoming calls and redirect them to the person caller wants. What I have so far searched from the voip-info.org and other sites, Ihave not found any example configuration how to do it. First i thought that I will implement this just using call parking,(operator just put them in hold) and person who call is meant, just picks it up.. But is this really the only way or is there some nicer way to handle that? Thank you in advance for your answers! This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to disconnect a bridged channel
Hi, i've just faced with some bridged calls which could not be hungup just killing the asterisk process solved the problem: Zap/63-1 (incoming s1 ) Up Bridged Call SIP/2035-e9cb logs say: Jul 22 14:54:12 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6785 seconds Jul 22 14:54:13 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6786 seconds ... warning:Jul 22 14:54:36 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! warning:Jul 22 14:54:36 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! warning:Jul 22 14:54:37 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! ... tones of these messages... I'm using latest CVS HEAD. thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marco and Realtime Extension Problem
Dear All,
I have a problem with the Marco and the Realtime Extensions in my
extensions.conf. The problem is that when I exit from my Marco, I
should return to my calling context, which is default but the next
step for it should be switch statement which will use realtime
extension. Somehow I am getting the following error below with
autofallthrough=yes :
-- Executing NoOp(SIP/555-5dcf, Channel is SIP/555-5dcf) in new stack
== Auto fallthrough, channel 'SIP/555-5dcf' status is 'UNKNOWN'
And the following error with autofallthrough=no :
-- Executing NoOp(SIP/555-f121, Channel is SIP/555-f121) in new stack
Jul 21 16:51:46 WARNING[4090]: pbx.c:2337 __ast_pbx_run: Timeout, but
no rule 't' in context 'default'
In a sense, when I leave the marco, I should be able to enter the
realtime extension, but the call flow just fails after prority of the
default context.
Is there some bug in my sytnax or something in the asterisk program itself?
Below is my default context:
[default]
exten = _X.,1,Macro(stdexten,${EXTEN},${CALLERIDNUM})
;Realtime Routing from MySQL
switch = Realtime/[EMAIL PROTECTED]
[macro-stdexten]
exten = s,1,NoOp(Leaving Marco)
Regards,
Kengie Ho
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Re: [Asterisk-Users] Re: Mahler's Book - New Project
Noah Miller wrote: snip In addition to largely being a rehash of existing docs on the internet, there are many editorial errors in the version that I have. Before I was comfortable with the conf files, these editorial errors were very confusing. The editions coming out now may have fixed these, but if not, it's just another reason to avoid thee book. I'd agree that the best way to get started is to get your hands wet. Be prepared to devote some time to learning asterisk. You'll find that in the end, it is still the quickest way, and well worth your effort. Someone told me of an O'Reilly book on Asterisk, and looking in their catalog i've found it: http://www.oreilly.com/catalog/asterisk/index.html Authors are credited as Jared Smith, Jim Van Meggelen and Leif Madsen, and it's due out in September '05. Has a picture of a starfish on the cover. Since it's not yet out, has anyone here proofread the thing, or has had an early copy, and willing to comment? Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] all zap channels get RING signal when starting *
hi all, when i start * all zap channels get ring signal so i get a huge number of incoming dummy calls when starting *. i'm using TE105P with 4 TA750 full fxo with latest CVS HEAD: zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsks=49-72 span=4,0,0,esf,b8zs fxsks=73-96 loadzone=us defaultzone=us zapata.conf: [channels] context=incoming callerid=asreceived busydetect=yes busycount=7 faxdetect=no signalling=fxs_ks overlapdial=no usecallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 channel = 1-96 thanks. Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming calls
hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg: Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??, thx in advance, CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de 1,49 euros par mois___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk captures sound device
On Fri, Jul 22, 2005 at 02:17:48PM +0400, Timur V. Elzhov wrote: When I run Asterisk (CVS HEAD version), I'm not able to play music anymore -- asterisk seems to capture sound device. Is it not a bug, but a feature? That's unlike stable (1.0.7 and 1.0.9) versions, when I can, say, run an IP telephone on the *same* machine and listen what Asterisk' autoattendant says. Now I can't do that, I need Asterisk and client running on the separate machines. Well, I've already reslolved that, by noload = chan_alsa.so noload = chan_oss.so in the `modules.conf'. -- Best regards, Timur Elzhov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found
hi Tzafrir, i was able to run make by removing ^M at the end of each line of each script, i also checked all script file on the /asterisk folder and execute dos2unix command on all script files, however when i run make i encountered another problem. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/in clude -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYD ETECT_MARTIN -fomit-frame-pointer-c -o md5.o md5.c md5.c: In function `byteReverse': md5.c:47: warning: cast increases required alignment of target type md5.c: In function `MD5Update': md5.c:98: warning: cast increases required alignment of target type md5.c:107: warning: cast increases required alignment of target type md5.c: In function `MD5Final': md5.c:142: warning: cast increases required alignment of target type md5.c:153: warning: cast increases required alignment of target type md5.c:154: warning: cast increases required alignment of target type md5.c:156: warning: cast increases required alignment of target type gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/in clude -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYD ETECT_MARTIN -fomit-frame-pointer-c -o term.o term.c In file included from include/asterisk/utils.h:26, from term.c:32: include/asterisk/strings.h:232: parse error before `va_list' include/asterisk/strings.h:232: warning: function declaration isn't a prototype make: *** [term.o] Error 1 bash-2.05# any ideas on how i can fix this? thnks in advance. chris. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queues and roundrobin/rrmemory
Round robin is designed to alternate between, in this case, the two agents. At least that is how I understand the comment in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Thursday, July 21, 2005 11:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] queues and roundrobin/rrmemory I have a queue setup using Asterisk CVS and roundrobin, however calls seem to be distributed in the same way as rrmemory (round robin with memory), ie, it is alternating between the two people in the queue rather than always calling the first available person in the queue first. I am using agents with agentcallbacklogin and addqueuemember to dynamically add the agent to the queue. asterisk version: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 2005-06-07 07:34:45 Does anyone use agents + agentcallbacklogin and use roundrobin queues with a recent CVS and have it working (or have the same problem ??) Thanks, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX over HTTP
On Fri, Jul 22, 2005 at 11:39:04AM +0200, Gustavo García wrote: You can traversal a HTTPS proxy using a plain TCP connection (without SSL). The unique requirement of some HTTPS proxys is that the target port is 443. Then if your Asterisk listen in 443 port IAX (TCP) connections, it should work. which makes this wishlist item a simple dependency of the wishlist item for IAX over TCP. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfert
David Romero a écrit : attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org http://voip-info.org have good example of features.conf On 7/21/05, *sylvain garcia* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David RomeroROMDAV ## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users tx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Hi, don't think it is the cable length because we tried to shorten it and nothing changed. We tried to use a Dell Poweredge with a TDM400P and a quadBRI using bristuffed-Asterisk 1.0.7 with no success...the only solution was removing tdm400P. We checked the interrupts but the two cards had their own. Couldn't be the server? The bad thing is BIOS doesn't allow to manually assign IRQs to the cards and it is hard to make tests. Is there anybody who had no problem configuring the cards above on the same server? How did you solve conflicts? TIA Giorgio David Hajek wrote: Thanks. My cable is like 8-10m long. Hm, will try to make shorter one but it works in old system. Who knows. - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com Michiel van Baak wrote: On 11:31, Wed 20 Jul 05, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Hi, I had the same errors too when I started to test with the 4port card. After changing the 200M UTP cable that was all put in a corner for a 2 meter cable the problems went away. I read on some previous posts from Klaus-Peter that the CRC errors mean bad cables. In my case the way-too-long cable from the NT1 to my * box was the cause. Maybe this can be of any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low profile FXO card
Hello,I am looking for 4 port FXO low profile PCI card that could be used with Asterisk.Digium TDM04B sound like a good choice but it is half high PCI card and I can not plug it in my Dell box (small box).I am looking for adequate low profile PCI card (55mm high or similar but definitely smaller than TDM04B so I can plug it in).Does anyone know where to search for it?Thank you in advance,Boris Zolotarev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi or mISDN for passive Fritz!Card PCi
Hi all, chan someone who has tried BOTH chan_capi and chan_mISDN with a passive Frtiz!Card PCI comment on one versus the other. Which had better sound quality. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX over HTTP
Doing IAX over TCP is simply a Bad Idea. Under perfect circumstances, it will work OK, but the slightest network disturbance will result in sound gaps/distortion and/or monster audio delay. This is not idle UDP-boosting, I've tried it. [Have had good results with UDP-based secure tunnel transport of IAX traffic (CIPE and OpenVPN)] On Fri, 22 Jul 2005, Tzafrir Cohen wrote: On Fri, Jul 22, 2005 at 11:39:04AM +0200, Gustavo García wrote: You can traversal a HTTPS proxy using a plain TCP connection (without SSL). The unique requirement of some HTTPS proxys is that the target port is 443. Then if your Asterisk listen in 443 port IAX (TCP) connections, it should work. which makes this wishlist item a simple dependency of the wishlist item for IAX over TCP.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk operator functions
[EMAIL PROTECTED] wrote: Hey! My asterisk is working properly so far with all automatic functions. Now I want to direct incoming calls to operator, i mean some person who answers to the [incoming] exten s,1,Answer() ; Wait for 2 seconds to pick up caller-id exten s,2,Wait(2) ; Dial operator's sip phone for 30 seconds exten s,3,Dial(SIP/4216,30,rt) ; If operator not found, play please wait exten s,4,Playback(find-operator) ; drop person into operator-group queue exten s,5,Queue(operator-group|t|||100) ; Hang up if they exit the queue before someone answers exten s,6,Hangup() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone have success with BRI in Italy?
Hi, the question is: can digium and quadBRI co-exists easily on the same server? We are still having a lot of troubles since it is hard to find infos on how to configure them. Giorgio. Emanuele Pucciarelli wrote: Kevin Hanson wrote: Can anyone recommend a BRI card that supports Asterisk and that will work in Italy? Will the Digium TDM card work in Italy? I guess that everyone here will recommend the quadBRI (e.g. Junghanns'). Digium's TDM card does not support BRI! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming calls
youa re using -v option multiple times at startup. That message is perfectly fine. ali kia wrote: hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg : Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... someone have an idea ??, thx in advance, http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link CaraMail met en oeuvre un nouveau *Concept de Sécurité Globale* http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link à partir de 1,49 euros par mois ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk operator functions
On Fri, Jul 22, 2005 at 01:43:01PM +0300, [EMAIL PROTECTED] wrote: Hey! My asterisk is working properly so far with all automatic functions. Now I want to direct incoming calls to operator, i mean some person who answers to the incoming calls and redirect them to the person caller wants. After a certain timeout, Asterisk will jump to extension 't' in the current context . So just like you can set up a default action using s, you can set up a timeout action using t. The operator can then transfer the calls. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfert
David Romero a crit: attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org have good example of features.conf On 7/21/05, sylvain garcia [EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Romero ROMDAV ## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users if i use asterisk 1.0.5 on debian attend transfert is present in feature.conf, but doesn't work? it's also for CVS head? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfert
sylvain garcia a crit: David Romero a crit : attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org http://voip-info.org have good example of features.conf On 7/21/05, *sylvain garcia* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David RomeroROMDAV ## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users tx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sorry i have 1.0.7 version it's possible of attended transfer? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] --- Problem with queues.conf and extensions.conf ---
Hi Asterisk-Users,
We have a problem with queues.conf / extensions.conf
queues.conf file reads like ...
member = SIP/8399
extensions.conf reads like ...
exten = 8399, 1, SetCIDNum(${AccountNumber}|a)
exten = 8399, 2, Dial(SIP/8399,10,Ttrf)
When somebody calls to the queue, we observed that
it is not going through extensions.conf
(previous two lines)
That mean's it is not executing dial plan rather it
is directly dialing to 8399.
We can observe this in asterisk-cmd-line where
SetCIDNum is not executed.
Anybody has some pointers on this problem, please
do reply.
Thanks in advance.
Best regards,
Somesh SS
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Re: [Asterisk-Users] Stupid hold music
Empire Records - Money (that's what I want) Erm, do you mean The Flying Lizards? My cousin was the vocalist on that one hit wonder. She makes a fotune out of XM Satellite Radio. They play it all the time on their 80's channel. Julien Goodwin wrote: On 22/07/2005 1:58 PM, Mark Phillips wrote: Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? Doesn't everybody, here's most of mine: (most shamelessly stolen from a discussion on a.s.r) Annie Lennox: Waiting in Vain Eurythmics: When Tomorrow Comes Moody Blues: Go Now PSB: Saturday Night Forever Pink Floyd: Time Tom Robinson: The Frozen Man Eurythmics: Forever Rolling Stones: Time Is On My Side Tommy Tutone - 867 5309 Kim Wilde - 36580 Blondie: Hanging on the Telephone ELO: Telephone Line Empire Records - Money (that's what I want) Pink Floyd: Money Blondie: Call Me ELO: Ma Ma Ma Belle And as background on the voice menus: Backman-Turner Overdrive: You Ain't Seen Nothing Yet Queen: I Want to Break Free Divine Comedy: The Certainty of Chance B-52's: 6068-842 Tom Robinson: 2-4-6-8 Motorway Queen: I'm going Slightly Mad Background for annoucements of queue position: Eurythmics: Would I Lie To You? Tom Lehrer: New Math ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP extension auto busy's itself
Hi Folks, I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until I recently upgraded to AAH from a rather old CVS build this was all working fine. He can call me in the other direction with no problems. Any ideas? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stupid hold music
On 22/07/2005 1:58 PM, Mark Phillips wrote: Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? My two (s)cents... * Anything from New Kids on The Block. * Put the pop-hook guitar riff from the Friends theme on repeat. * The Happy Flowers (http://members.aol.com/_ht_a/MrHCIHF/ , http://www.mp3.com/happy-flowers/artists/9826/summary.html ) have a song called Mom, I Gave the Cat Some Acid. * If you can get the song from this flash animation converted to MP3, then it might be good (bad): http://www.ebaumsworld.com/flash/spacepeople.html . * Just do a web search for Industrial Sound Collage. I'm sure you'll find something appropriate. * I have a Barney-themed sound collage I created that I might be willing to share. Hope this helps (hinders?) your business. - Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stupid hold music
Sunshine, lolly-pops, and rainbows; everything that's wonderful is what I feel when we're together ... Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Melanson Sent: Friday, July 22, 2005 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stupid hold music On 22/07/2005 1:58 PM, Mark Phillips wrote: Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? My two (s)cents... * Anything from New Kids on The Block. * Put the pop-hook guitar riff from the Friends theme on repeat. * The Happy Flowers (http://members.aol.com/_ht_a/MrHCIHF/ , http://www.mp3.com/happy-flowers/artists/9826/summary.html ) have a song called Mom, I Gave the Cat Some Acid. * If you can get the song from this flash animation converted to MP3, then it might be good (bad): http://www.ebaumsworld.com/flash/spacepeople.html . * Just do a web search for Industrial Sound Collage. I'm sure you'll find something appropriate. * I have a Barney-themed sound collage I created that I might be willing to share. Hope this helps (hinders?) your business. - Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP extension auto busy's itself
I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until I recently upgraded to AAH from a rather old CVS build this was all working fine. He can call me in the other direction with no problems. Any ideas? Its almost impossible to guess at a problem without some hints from your systems. I good start might be the CLI results from both systems when good and bad calls are made. Without that info, a wild ass guess is that WiSIP phone is not properly registering with asterisk, or, the registration is timing out, or, something like that. When the call to the WiSIP phone fails, what does 'sip show peers' display? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Quadbri trouble
I got the same problem, and what i change it was point-to-point to point-to-multi-point, and I used normal patch cable to connect the ISDN card to ISDN connector, and I used TE mode. In italy work like that I hope that will help you, I used also bristuff. Good luck Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED] Inviato: martedì 19 luglio 2005 16.23 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Quadbri trouble Hi, I'm trying to configure a quadbri card using the configuration found in Bristuff. I know the configuration of telco is point-to-point and I think the card have to work in NT mode (I presume because I have not found the documentation about this and when attach to the ISDN the led become green). I'm not able to make and receive call. When I receive a call nothing happen in the asterisk cli and from the caller I receive a connection error. When I try to make a call I receive unable to create zap channel THis is my configuration: Zapata.conf switchtype = euroisdn signalling = bri_net pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes context=default group = 1 ; S/T port 1 channel = 1-2 signalling = bri_net group = 2 ; S/T port 2 channel = 4-5 signalling = bri_net group = 3 ; S/T port 3 channel = 7-8 signalling = bri_net group = 4 ; S/T port 4 channel = 10-11 signalling = bri_net Zaptel.conf loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stupid hold music
Oh man, they are just plain B, not even worth viewing which is a shame because this mp3 site looked promising. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeremy Melanson Sent: Friday, 22 July 2005 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stupid hold music On 22/07/2005 1:58 PM, Mark Phillips wrote: Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? My two (s)cents... * Anything from New Kids on The Block. * Put the pop-hook guitar riff from the Friends theme on repeat. * The Happy Flowers (http://members.aol.com/_ht_a/MrHCIHF/ , http://www.mp3.com/happy-flowers/artists/9826/summary.html ) have a song called Mom, I Gave the Cat Some Acid. * If you can get the song from this flash animation converted to MP3, then it might be good (bad): http://www.ebaumsworld.com/flash/spacepeople.html . * Just do a web search for Industrial Sound Collage. I'm sure you'll find something appropriate. * I have a Barney-themed sound collage I created that I might be willing to share. Hope this helps (hinders?) your business. - Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
I found the same problem, u have to use normal patch cable to connect from ISDN card to ISDN connector, And u r in italy I think, in italy u have to use p2mp TE mode signalling=bri_cpe_ptmp, with telcom, Good luck -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Marc Sutter Inviato: lunedì 18 luglio 2005 18.58 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received there is a list item about this issu: http://lists.digium.com/pipermail/asterisk-users/2004-November/070918.html alternatively you can try this solution on debian: # 4)Modules configuration for startup with zaphfc AND wcfxs on # debian stable 3.1 sarge kernel 2.4.27-2-386 # Here a configuration to fix this issue at boottime emacs /etc/modutils/zaptel # match it to: post-install zaphfc /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcusb /sbin/ztcfg #post-install wcfxo /sbin/ztcfg #post-install ztdynamic /sbin/ztcfg #post-install ztd-eth /sbin/ztcfg #post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg #post-install wcte11xp /sbin/ztcfg alias wctdm wcfxs #post-install torisa /sbin/ztcfg #post-install wcfxs /sbin/ztcfg # end of file /etc/modutils/zaptel #Update the /etc/modules.conf file with: [EMAIL PROTECTED] update-modules [EMAIL PROTECTED] emacs /etc/modules # add a line like the following at end of the file zaphfc # and finaly reboot for testing [EMAIL PROTECTED] reboot On jeu, 2005-07-07 at 18:18 +0200, Yousef Herzallah wrote: I have this problem zaphfc: empty HDLC frame or bad CRC received My configurations are cat /proc/zaptel/1 Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3) AMI/CCS 1 ZTHFC1/0/1 Clear 2 ZTHFC1/0/2 Clear 3 ZTHFC1/0/3 HDLCFCS cat /etc/zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] language=it switchtype=euroisdn ; p2mp TE mode ;signalling=bri_cpe_ptmp ; p2p TE mode ;signalling=bri_cpe ; p2mp NT mode ;signalling=bri_net_ptmp ; p2p NT mode signalling=bri_net pridialplan=dynamic prilocaldialplan=local nationalprefix=0 internationalprefix=00 echocancel=yes echotraining=100 echocancelwhenbridged=yes immediate=yes group=1 context=default channel = 1 channel = 2 ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] --- Problem with queues.conf and extensions.conf ---
Hi Patrick,
Removing spaces didn't help in this regard.
Some other solutions?
Best regards
Somesh SS
--- Patrick [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 05:38 -0700, somesh s wrote:
exten = 8399, 1, SetCIDNum(${AccountNumber}|a)
exten = 8399, 2, Dial(SIP/8399,10,Ttrf)
^
I think you need to remove the remove spaces.
so:
exten = 8399,1,SetCIDNum...
exten = 8399,2,Dial
Regards,
Patrick
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Re: [Asterisk-Users] IAX over HTTP
Not that it need any additional 'push' against it, :-).. My tests with IAX over OPENVPN (on port 443) are acceptable (they do work just fine) for basic non-user-friendly purposes. Examples, I get my voice mail at home sometimes via this tunnel (if wife using primary landline. I test my dialplan via firefly over this tunnel. I called my family over FWD but really nothing to be used for anything that really matter. Would not pass any non-geek acceptance test. Jerry Glomph Black wrote: Doing IAX over TCP is simply a Bad Idea. Under perfect circumstances, it will work OK, but the slightest network disturbance will result in sound gaps/distortion and/or monster audio delay. This is not idle UDP-boosting, I've tried it. [Have had good results with UDP-based secure tunnel transport of IAX traffic (CIPE and OpenVPN)] On Fri, 22 Jul 2005, Tzafrir Cohen wrote: On Fri, Jul 22, 2005 at 11:39:04AM +0200, Gustavo García wrote: You can traversal a HTTPS proxy using a plain TCP connection (without SSL). The unique requirement of some HTTPS proxys is that the target port is 443. Then if your Asterisk listen in 443 port IAX (TCP) connections, it should work. which makes this wishlist item a simple dependency of the wishlist item for IAX over TCP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell Hardware
Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stupid hold music
Happy Tree Friends' theme is all you need to annoy who's on hold (even if actually I don't know if you can use it for business purpose) Anyway, isn't time to split this list in strictly technical questions-asterisk-users and what's the best provider/hardware/moh/book/distro/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users lists ? Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interconnect with Mitel PBX
I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or what other way could it work? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
I am observing. The problem is in the outbound calls. Some are not completed. Thank you - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning never connected or connected then disconnected abruptly? Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 17:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel = 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
On Friday 22 July 2005 14:48, Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? http://www.digium.com/index.php?menu=compatibility Digium's recommendation is quite clear: 'Don't use Dell hardware' And it's a great shame Digium hardware has such problems on Dell kit, since there's so much of it about :( Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
Il giorno ven, 22/07/2005 alle 08.48 -0500, Anton Krall ha scritto: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? I'm using a DELL PE750 server with an AVM c2, suse-pro installed, capi works out of the box. DELL servers do a lot of noise, consider it if you aren't putting the server in a dedicated room, really ... I have two of them in the corridor out of my office, they drive me insane ... Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Norstar MICS
To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. Norstar1 PRI Asterisk-1 IP-WAN Asterisk-2 ---PRI--- Norstar2 The Norstars are MICS 0x32 4.1 software Thanks in advance Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
Hi there, Our problem is with outgoing calls... And the problem is some calls do not complete...the asterisk show the ring...but doesnt complete some calls...we dont have dropped calls... thank you - Original Message - From: Paul Belanger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 21, 2005 6:45 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls Are your problems with incoming calls to your PRI or outgoing calls? Are the calls being dropped or just not hitting your asterisk box? PB JOAO CARLOS MOURA wrote: Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel = 1-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] YAACID - 0.91 new release
YAACID 0.91 has been released. You can access it on the web site http://www.shatterit.com/opensource/yaacid this should fix some problems with [EMAIL PROTECTED] and older cvs and non-cvs asterisk versions. (the manager interface has changed quite a bit, which was causing the problems) Theres also an advanced configuration for those that know what they are doing..the documentation has not been updated yet, but will be shortly. Please feel free to contact me with any questions. Best Regards, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low profile FXO card
Boris Zolotarev - Pamet wrote: Digium TDM04B sound like a good choice but it is half high PCI card and I can not plug it in my Dell box (small box). I am looking for adequate low profile PCI card (55mm high or similar but definitely smaller than TDM04B so I can plug it in). You will not find one, I'd say. There is not enough room on a low-profile PCI card bracket for four RJ11 connectors, and using any other connectors would make the board hard to get certified. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ Bruno De Luca escribió: We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stupid hold music
Simone Cittadini wrote: snip Anyway, isn't time to split this list in strictly technical questions-asterisk-users and what's the best provider/hardware/moh/book/distro/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users lists ? Simone Cittadini IT Manager And lest we forget, another split for the Cisco/Polycom/Snom/othersipphone configurations The list police complain about questions on AAH and AMP which are more closely related than these. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
[EMAIL PROTECTED] wrote: And it's a great shame Digium hardware has such problems on Dell kit, since there's so much of it about :( If you don't use digium hardware, there's of course no problems with using Dell. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stupid hold music
There are 10 types of people in the world: those who understand binary and those who don't. why are these stupid quotes so amuzing? love it!!On 22/07/05, Thomas Christie [EMAIL PROTECTED] wrote: Sunshine, lolly-pops, and rainbows; everything that's wonderful is what Ifeel when we're together ...Thomas ChristieThere are 10 types of people in the world:those who understand binary andthose who don't. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of JeremyMelansonSent: Friday, July 22, 2005 09:08To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Stupid hold musicOn 22/07/2005 1:58 PM, Mark Phillips wrote: Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc?My two (s)cents...* Anything from New Kids on The Block. * Put the pop-hook guitar riff from the Friends theme on repeat.* The Happy Flowers (http://members.aol.com/_ht_a/MrHCIHF/ , http://www.mp3.com/happy-flowers/artists/9826/summary.html ) have a songcalled Mom, I Gave the Cat Some Acid.* If you can get the song from this flash animation converted to MP3, thenit might be good (bad): http://www.ebaumsworld.com/flash/spacepeople.html .* Just do a web search for Industrial Sound Collage. I'm sure you'll findsomething appropriate. * I have a Barney-themed sound collage I created that I might be willingto share.Hope this helps (hinders?) your business.-Jeremy___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
pri show span 1 Primary D-channel: 24Status: Provisioned, Up, ActiveSwitchtype: National ISDNType: CPEWindow Length: 0/7Sentrej: 0SolicitFbit: 0Retrans: 0Busy: 0Overlap Dial: 0T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer: 4000T313 Timer: 4000N200 Counter: 3 thks - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURASent: Thursday, July 21, 2005 17:56To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel = 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
My debugThank you for help.
Verbosity is at least 5 --
Accepting AUTHENTICATED call from
requested format = g729, requested
prefs = (), actual format =
gsm, host prefs =
(gsm), priority =
mine -- Executing AbsoluteTimeout("IAX2/[EMAIL PROTECTED]", "3600") in new
stack -- Set Absolute Timeout to
3600 -- Executing SetCallerID("IAX2/[EMAIL PROTECTED]", "9545569050") in new
stack -- Executing Ringing("IAX2/[EMAIL PROTECTED]", "") in new
stack -- Executing Dial("IAX2/[EMAIL PROTECTED]",
"ZAP/g1/0115491140583282|60|tr") in new stack-- Making new call for cr
42038 -- Requested transfer capability: 0x00 -
SPEECH Protocol Discriminator: Q.931 (8) len=52 Call Ref:
len= 2 (reference 9270/0x2436) (Originator) Message type: SETUP
(5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1
Q.931 Std: 0 Info transfer capability: Speech
(0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83
81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User
(0)
Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ]
[6c 0c 21 81 39 35 34 35 35 36 39 30 35 30] Calling Number (len=14) [
Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163)
(1)
Presentation: Presentation permitted, user number passed network screening (1)
'9545569050' ] [70 11 a1 30 31 31 35 34 39 31 31 34 30 35 38 33 32 38
32] Called Number (len=19) [ Ext: 1 TON: National Number (2)
NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0115491140583282'
] -- Called g1/0115491140583282 Protocol
Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference
9270/0x2436) (Terminator) Message type: CALL PROCEEDING (2) [18
03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI
Spare: 0, Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel
Identification) Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 9270/0x2436) (Terminator) Message type:
PROGRESS (3) [1e 02 8a 81] Progress Indicator (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network
beyond the interworking point
(10)
Ext: 1 Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]-- Processing IE 30 (cs0,
Progress Indicator)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing
call Proceeding, peerstate Incoming Call Proceeding Protocol
Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference
9270/0x2436) (Originator) Message type: DISCONNECT (69) [08 02
81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
] -- Hungup 'Zap/1-1' == Spawn extension (qvox,
0115491140583282, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2
(reference 9270/0x2436) (Terminator) Message type: RELEASE
(77)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request Protocol Discriminator: Q.931 (8) len=9 Call Ref:
len= 2 (reference 9270/0x2436) (Originator) Message type: RELEASE
COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network
serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate
Null
- Original Message -
From:
Steve Totaro
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, July 21, 2005 11:33
PM
Subject: Re: [Asterisk-Users] T1 -
incomplete calls
pri debug span 1 output?
- Original Message -
From:
Thomas Christie
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, July 21, 2005 4:14
PM
Subject: RE: [Asterisk-Users] T1 -
incomplete calls
Incomplete meaning "never connected" or "connected then
disconnected abruptly?" Are the calls inbound or outbound? All
calls or just some calls? If just some, about what percentage are
problem calls?
Try setting Switchtype = 5ess, 4ess, etc. Let me
know what you notice, if anything is different.
Thomas Christie
There are 10 types of people in the
world: those who understand binary and those who don't.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JOAO
CARLOS MOURASent: Thursday, July 21, 2005 17:56To:
Re: [Asterisk-Users] Asterisk and Norstar MICS
On Friday 22 July 2005 10:15, Michael Di Martino wrote: My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. You need to tell the norstar that you are done dialing. It's waiting for more digits. Routing Service, Public DN Lengths and adjust the correct prefix. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uk Caller id
Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen 0 (-20)Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: SuccessJul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels]context=defaultswitchtype=nationalsignalling=fxo_lsrxwink=300 ; Atlas seems to use long (250ms) winksusecallerid=ukcallerid=asreceivedcidsignalling=v23cidstart=usehistcallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesgroup=1callgroup=1pickupgroup=1immediate=noprogzone=ukmusiconhold=default ; incoming channels signalling=fxs_ksgroup=2context=incomingchannel = 1-2 ; outgoing channels signalling=fxo_ksgroup=1context=outgoingchannel = 3 Thanks loads Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
It would be helpful to capture a complete ISDN call setup. On the cli type pri debug span 1 Then place a call and turn off debug with pri no debug span 1 You will then have a complete listing of the signalling between your co and your * for this time period. Good Luck On Jul 22, 2005, at 9:34 AM, JOAO CARLOS MOURA wrote: pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 thks - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning never connected or connected then disconnected abruptly? Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 17:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel = 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
I see the Dell SC420 is discarded according to Digium but what about the SC430, SC1420 or others? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Simone Cittadini |Sent: Viernes, 22 de Julio de 2005 09:12 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Dell Hardware | |Il giorno ven, 22/07/2005 alle 08.48 -0500, Anton Krall ha scritto: | Guys. | | What do you think about Dell hardware and Asterisk? Whos using it, | comments, any special specs recommended or models? | |I'm using a DELL PE750 server with an AVM c2, suse-pro |installed, capi works out of the box. DELL servers do a lot of |noise, consider it if you aren't putting the server in a |dedicated room, really ... I have two of them in the corridor |out of my office, they drive me insane ... | |Simone Cittadini |IT Manager |== |COMVERT S.R.L. |via F.lli Bressan, 21 |20126 Milano - ITALY |Tel +39.02.27006796(aspetta un beep)105 |[EMAIL PROTECTED] |http://www.comvert.com | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stupid hold music
I was thing about XTCs stupidly happy M. On Fri, 22 Jul 2005 15:57:07 +0200, Simone Cittadini wrote: Happy Tree Friends' theme is all you need to annoy who's on hold (even if actually I don't know if you can use it for business purpose) Anyway, isn't time to split this list in strictly technical questions-asterisk-users and what's the best provider/hardware/moh/book/distro/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users lists ? Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
Was this an exmple of your incomplete calls? From the trace it appears that you issued the disconnect while the call was in process. On Jul 22, 2005, at 9:32 AM, JOAO CARLOS MOURA wrote: My debug Thank you for help. Verbosity is at least 5 -- Accepting AUTHENTICATED call from requested format = g729, requested prefs = (), actual format = gsm, host prefs = (gsm), priority = mine -- Executing AbsoluteTimeout(IAX2/[EMAIL PROTECTED], 3600) in new stack -- Set Absolute Timeout to 3600 -- Executing SetCallerID(IAX2/[EMAIL PROTECTED], 9545569050) in new stack -- Executing Ringing(IAX2/[EMAIL PROTECTED], ) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED], ZAP/ g1/0115491140583282|60|tr) in new stack -- Making new call for cr 42038 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=52 Call Ref: len= 2 (reference 9270/0x2436) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u- Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 21 81 39 35 34 35 35 36 39 30 35 30] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '9545569050' ] [70 11 a1 30 31 31 35 34 39 31 31 34 30 35 38 33 32 38 32] Called Number (len=19) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0115491140583282' ] -- Called g1/0115491140583282 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 9270/0x2436) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 9270/0x2436) (Terminator) Message type: PROGRESS (3) [1e 02 8a 81] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] -- Processing IE 30 (cs0, Progress Indicator) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing call Proceeding, peerstate Incoming Call Proceeding Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 9270/0x2436) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Spawn extension (qvox, 0115491140583282, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]' -- Hungup 'IAX2/[EMAIL PROTECTED]' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 9270/0x2436) (Terminator) Message type: RELEASE (77) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 9270/0x2436) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning never connected or connected
[Asterisk-Users] zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
Hi, I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs modules and everything seemed allright but linux log shows the following message: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. Anybody knows what it means? TIA Giorgio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] attended transfert
sorry i have 1.0.7 version it's possible of attended transfer? No, it is only available in CVS. Udo PS: Please don't quote all of the other messges you are replying to. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WAS: Stupid hold music NOW: list gripes
Anyway, isn't time to split this list in strictly technical... And lest we forget, another split for the Cisco/Polycom/Snom/othersipphone configurations... That would be short-sighted imo. Splitting the list or goofy offers to do the list as a PHPbb, NNTP, or other forums would only serve to dilute the value of the collective wisdom of the people on this list. Suppose the lists changed to -users, -newbies, -sip, -interop, -biz, -dev, -phones. I have a question about pushing a firmware upgrade to a Snom 190 (I did last week, thank to all that helped). Now, what list do I post it to? -users? -sip? -interop? -phones? What if the Snom gentleman that helped me out last week was on -phones only, but I posted to -interop? The answer to my question was so close but I had to divine that the source of the information that I needed was on a different list. So I crosspost. Some guys are like me, they subscribe to all the lists, so they see the same question X 7 times. So exactly, how did splitting the list help reduce the traffic or focus the topic? Take the lessons from Usenet. There will be 20 different NG's all on the same general topic. Using the Microsoft forums for example, you will see microsoft.sqlserver.programming, microsoft.sqlserver.general, microsoft.sqlserver.questions etc. There is so litle stratification between topics and so many topics crossover to different topics, that everyone just gives up and posts to microsoft.sqlserver.general anyway, and the other NG's atrophy, to the point where literally 90% of all existing newsgroups could be deleted today, and no one would miss out (except the spammers). Same thing I see here if the list gets split. No one would post to -newbies, -sip, -interop, -phones, they'd all post to -users 'cause they know all the l33t guys subscribe to that list. I for one am immensely grateful that this list is heavily traffic'd with OT, flames, anecdotes and the like. It demonstrates that this is a living, vibrant list with active contributors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
Kevin Walsh wrote: The perpetual agreement grants the owner a non-cancellable right to use changes and/or enhancements made to the Asterisk codebase as [the] owner sees fit. As any Asterisk fork would, of course, be based upon existing Asterisk code, the owner would have the automatic right to take any code they wanted and backport it into the Asterisk Binary Edition - as long as the contributor to the fork had previously signed a perpetual disclaimer at some point in the past. Nice work clipping out only the words you wanted to use there! Let's try this again, with the actual text from the disclaimer: (b) The rights made in Para. 1(a) of this Agreement applies to all past and future contributions of Contributer that constitute changes and enhancements to the Program. 2. Contributer shall report to Owner all changes and/or enhancements to the Program which are covered by this Agreement, and (to the extent known to Contributer) any outstanding rights, or claims of rights, of any person, that might be adverse to the rights of Contributer or Owner. In other words, the _only_ code that the disclaimer covers is that which the Contributer directly identifies to Digium to be covered by the disclaimer. In absolutely no way does this disclaimer give Digium the right to appropriate other changes the Contributer makes to the covered programs without their knowledge and permission. In addition, even the most liberal interpretation of these clauses still includes the words Contributer and contribution, which clearly means that the entity signing the disclaimer has sole discretion which of their changes are covered and which are not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uk Caller id
this is an italian code and works... try it. [channels] ; -- canale 4 -- language=en faxdetect=both musiconhold=default group=2 canpark=yes context=inbound signalling=fxs_ks usecallerid=no ; echo cancel echocancel=128 ; range from 32 to 256(=echo 100%) echocancelwhenbridged=yes ; yes = 400 msec echotraining=200 channel=4 Bruno De Luca Graziosi Chris Thompson wrote: Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen 0 (-20) Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: Success Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=uk callerid=asreceived cidsignalling=v23 cidstart=usehist callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no progzone=uk musiconhold=default ; incoming channels signalling=fxs_ks group=2 context=incoming channel = 1-2 ; outgoing channels signalling=fxo_ks group=1 context=outgoing channel = 3 Thanks loads Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
I'm using a Dell GX270 with a single TE110P, no problems here. Of course I had to take off the pci aluminum card holder thingy to fit in the half height case, but it works great. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thibault Lamy Sent: Friday, July 22, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Hardware We are using a Dell PE SC1420 as asterisk server with one Beronet QuadBRI card (bristuff) and one Digium TE110p and it works well. No IRQ conflict. Thib. Elio Rojano wrote: DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ Bruno De Luca escribió: We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
What do you recommend for E1 and Analog (ala TDM400p)? Have you tested Dell with other cards yourself? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Andreas Sikkema |Sent: Viernes, 22 de Julio de 2005 09:37 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Dell Hardware | |[EMAIL PROTECTED] wrote: | | And it's a great shame Digium hardware has such problems on |Dell kit, | since there's so much of it about :( | |If you don't use digium hardware, there's of course no |problems with using Dell. | |-- |Andreas Sikkema bbned NV |Van Vollenhovenstraat 33016 BE Rotterdam |t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
We are using a Dell PE SC1420 as asterisk server with one Beronet QuadBRI card (bristuff) and one Digium TE110p and it works well. No IRQ conflict. Thib. Elio Rojano wrote: DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ Bruno De Luca escribió: We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RE: Business Edition
Aidan Van Dyk wrote: So what are they planning on doing with the Google Summer of Code results? http://code.google.com/summfaq.html#what_licenses_will_i_have What licenses will I have to choose from? This depends on your mentoring organization. For instance if Google is your mentoring organization, we will require you to choose either the BSD (sans advertising), LGPL or the GPL license for your project. http://code.google.com/summfaq.html#who_owns_the_software_i_w Who owns the software I write? You or your mentoring organization must license your code under a license palatable to your mentoring organization. Some organizations will require you to assign copyright to them, but many will allow you to retain copyright. If Google is your sponsoring organization, then you keep the copyright to your code. Did they really sign up as a mentor just to get the 500 bucks? It's truly amazing how 30 seconds of reading is more productive than spreading libelous FUD. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
Juraj Bednar wrote: Hello, I just got my Soekris 4801 box for use with Asterisk, but not as a primary Asterisk server. * [EMAIL PROTECTED] (Is @home or regular better?) If you want to run from CF, I recommend running some distribution (that does not take much space) and your own Asterisk... I'm not even sure if it be that easy to install Asterisk on Soekris in the first place. It should take less than five minutes: http://www.astlinux.org I found documentation not being that good for installs, I found a wonderful page describing the install process: http://www.ultradesic.com/index.php?section=22 * Shorwall firewall Try to get a real firewall, Shorewall has quite high latency. You should optimize... Shorewall is an abstraction layer for iptables. I don't know what you mean by real firewall but iptables does the work here. It most certainly is a real firewall. Shorewall probably has high latency because it adds a lot of frivolous rules (just like every other firewall utility of it's kind). What is the CF size you are using? and how much is still free? What have you installed? For my setup I installed OpenBSD, although I primarily use Debian GNU/Linux. The OpenBSD choice was because of the vpn card for Soekris, which is better supported under OpenBSD. I installed the base package except games and manual pages, about 60MB was still free (I used 256MB compact flash card). You can use the flashboot script to generate a nice trimmed down OpenBSD environment... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
I know. but u can't disable the USB controller always. If u have an server w/ others functions... Bruno De Luca Graziosi DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Hi, I have also failed the same point. Mine is 5.4-Stable Jul 16, did make world from 5.3 which works * 1.0.6(?) ports and I did cvsup ports-supfile again several minutes ago. NG. -- Zen Darren Wiebe wrote Did you do a make clean? I just, as in 1 hour ago, successfully installed 1.0.9 using the port on FreeBSD. Yeah, even deleted all the files in the asterisk ports , and refreshed it ports collection. Always fails to compile at this point. Am I missing a package dependency somewhere? Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Advice
I need to place a SIP FXO gateway in Central America. I've been looking at Quintum products, but the prices are about $150/FXO port. I have a Dell SC400 on the shelf, and I'm considering just installing Asterisk and two TDM04B cards and shipping it down. Does anyone have multiple TDM cards in the SC400? FXO ports on a TDM card are about $75/FXO port. With the Dell, there will be the advantage of trunked IAX (we're paying for bandwidth on both ends). Can anyone tell me if I'm missing something. The mains are 110V. What about answer and disconnect supervision (I'm assuming there isn't any on the FXO circuits). Is there some other way to do this for around $500? Thanks for your help, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAACID - 0.91 new release
Where is the source code? It looks interesting, But I´m interested in modifying the app, so, only with a administration password o even a file, you can configure the app... Users are quite curious about menus and always are looking to improve de functionality of the configured software... And if there is no source code, its ok :) then is just my 2 cents. Mark Musone wrote: YAACID 0.91 has been released. You can access it on the web site http://www.shatterit.com/opensource/yaacid this should fix some problems with [EMAIL PROTECTED] and older cvs and non-cvs asterisk versions. (the manager interface has changed quite a bit, which was causing the problems) Theres also an advanced configuration for those that know what they are doing..the documentation has not been updated yet, but will be shortly. Please feel free to contact me with any questions. Best Regards, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue issues: timeout and leastrecent strategy
Joerg, You will need to have the agent kicked from the queue for it to properly pass to the next agent.. In agents.conf under [agents], set autologoff=10. This will kick them out of the queue(logs them out) for not answering after 10 seconds. --johann Joerg Wolf wrote: Hi, I've configured a queue with dynamic agents and leastrecent strategy. If the least recent agent doesn't pick up the current call from the queue, the call will be presented to him again and again, even when there's yet another agent available. I would expect that after timeout occurs on the first agent, the next to least recent agent will be tried and so on and so forth... (as it happens in case of an busy least recent agent). Did I miss something in the config or is this the intended behaviour? Thanks! cheers Jörg Nutzen Sie Ihr Postfach als virtuelle Festplatte! - Jetzt installieren! http://mail.lycos.de/app/lycosinside/setupLI.exe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RE: Business Edition
Aidan Van Dyk wrote: Kevin P. Fleming wrote: The first two statements are true; the third is not. While you can certainly distribute the code you contribute to us via any other means you wish (under any other license you wish, including the GPL), the Digium Asterisk source tree cannot accept GPL only code. So what are they planning on doing with the Google Summer of Code results? The Summer of Code licensing requirements are sufficiently ambiguous as to allow the Asterisk work to be licensed as public domain, and that would suit Digium just fine, I suspect. Did they really sign up as a mentor just to get the 500 bucks? Well, I'm sure that was an added bonus. :-) Free work and free money. It reminds me of a certain Dire Straits lyric. Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
Kevin P. Fleming wrote: Kevin Walsh wrote: The perpetual agreement grants the owner a non-cancellable right to use changes and/or enhancements made to the Asterisk codebase as [the] owner sees fit. As any Asterisk fork would, of course, be based upon existing Asterisk code, the owner would have the automatic right to take any code they wanted and backport it into the Asterisk Binary Edition - as long as the contributor to the fork had previously signed a perpetual disclaimer at some point in the past. Nice work clipping out only the words you wanted to use there! Let's try this again, with the actual text from the disclaimer: Aw Kevin that's no fun; it's more fun to poke up trouble and try to turn people against Digium. Kevin Walsh and Aidan are able to see things that the rest of us cannot. Digium has duped you into associating with their evil enterprise to appropriate everyone else's hard work. I'm sure the stuff you and Mark have contributed pales in comparison with *their* contributions!! b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no active channel but one active call???
hi, what does this mean?: www*CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channels 1 active call after some searchs got this: www*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 172.22.22.27 239920830697669 00101/3343865 ulaw Rx: ACK 1 active SIP channel(s) logs say: Jul 22 20:28:59 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:00 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8106 seconds Jul 22 20:29:00 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:01 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8107 seconds Jul 22 20:29:01 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:02 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8108 seconds Jul 22 20:29:02 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:02 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8108 seconds thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] web managment
what is the best web based managment aplication for asterisk ??? Dante ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAACID - 0.91 new release
i'l lhave the source code posted to the site tomorrow..all day yesterday and today, we've been making major changes from the feedback from others... The asterisk manager interface is killing us! every version of asterisk outputs different events and information..so we're trying to see all the different events that indicate an incoming call..what a pain!! -Mark On 7/22/05, Andres Tello Abrego [EMAIL PROTECTED] wrote: Where is the source code? It looks interesting, But I´m interested in modifying the app, so, only with a administration password o even a file, you can configure the app... Users are quite curious about menus and always are looking to improve de functionality of the configured software... And if there is no source code, its ok :) then is just my 2 cents. Mark Musone wrote: YAACID 0.91 has been released. You can access it on the web site http://www.shatterit.com/opensource/yaacid this should fix some problems with [EMAIL PROTECTED] and older cvs and non-cvs asterisk versions. (the manager interface has changed quite a bit, which was causing the problems) Theres also an advanced configuration for those that know what they are doing..the documentation has not been updated yet, but will be shortly. Please feel free to contact me with any questions. Best Regards, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set the SMSC sender = VoIP provider 10-digit #
I have a VoIP provider that terminates to a PSTN 10-digit number. I want to be able to send/receive SMS messages that appear to be from the 10-digit number that my VoIP provider gives me. I am currently integrating Asterisk with Kannel and it works great. Unfortunately, my mobile phone provider Cingular, doesn't allow me to alter my number when I send out SMS messages. Or at least I haven't gotten it working. I am looking for an *affordable* solution that will give me access to the carrier network and allow me to set my SMS number so it matches the VoIP PSTN terminated one. Does anyone have any suggestions for me? Hope this gives you enough information. I recognize the term affordable is rather subjective. Regards, HJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
