Re: [asterisk-users] How to make call from asterisk?
Hi Neoh, All you have to do is configure your VoIp provider as another SIP extension on your Asterisk server and then use extensions.conf to set dialout rules, so when you do dial a number your asterisk server forwards it to the VoIp provider. Examples of extensions.conf can be found at http://www.asteriskguru.com/tutorials/extensions_conf.html http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf I can send you my extensions.conf if you want a working example. I do something very similar with a VoIP provider that provides an SIP interface. Hope this helps. On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, Thanks for your reply.. I am intend to dial using a VOIP provider.(developed by us) Software: x-Lite (SIP softphone) Registration of account number is fine, but for the case when i dial a number, it prompt out a message that the number not found. From my understanding, asterisk can be SIP server? or we need to implement a SIP server to integrate with Asterisk in order to provide full picture of VOIP system? Thanks. Date: Wed, 5 Sep 2007 13:30:21 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to make call from asterisk? Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Live Search: Better results, fast Try it now! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing call restriction
Dear all I want to restrict outgoing call from specified extention so is there any configuration for this setup ?? please send me example file - Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX with asterisk
Dear all I have fax machine which is connected with audiocode FXS port and audiocode connected with my asterisk server now what configuration i have to configured on asterisk ?? can any one suggest me what would be best for this kind of setup ?? [FAX]--[Audiocode][asterisk]--[PRI] - Got a little couch potato? Check out fun summer activities for kids.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with calling queues
Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP handsets, one VOIP account with a telco and one PSTN. The server correctly allows: - Handsets to call each other - Calls outbound through both PSTN or VOIP I'm having an issue with incoming calls however. If I configure incoming calls coming over my PSTN to a single user, it works correctly (that handset rings, can pickup etc). However if I define a call queue which consists of both these handsets, neither ever rings. Looking at the console, I see this: -- Started music on hold, class 'default', on Zap/1-1 [Sep 6 02:22:51] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: Unexpected control subclass '2' [Sep 6 02:22:54] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: Unexpected control subclass '2' The error repeats until the caller hangs up. I've posted all the config that I felt was relevant here, let me know if you need more. This was all written by Asterisk-GUI. I realise there's a lot more configuration but given that things work fine when I set the receive to a single agent, I assumed it was a queue issue. Users.conf [6001] callwaiting = yes context = numberplan-custom-1 email = [EMAIL PROTECTED] fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = SECRET threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 Queues.conf [6003] fullname = All of us strategy = ringall timeout = wrapuptime = autofill = yes autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = member = Agent/6001 member = Agent/6002 extensions.conf - broken [DID_trunk_2] include = default exten = _X.,1,Goto(default|6003|1) exten = s,1,Goto(default|6003|1) extensions.conf - works but only sends to a single handset [DID_trunk_2] include = default exten = _X.,1,Goto(default|6001|1) exten = s,1,Goto(default|6001|1) Any assistance appreciated. Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 | www.visinet.com.au http://www.visinet.com.au/ This e-mail is intended for use by the named recipients only and contains confidential information. Opinions and other information in this message that pertain to the sender's employer and its products and services represent the opinion of the sender and not necessarily those of the employer. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
On Wed, Sep 05, 2007 at 01:36:02AM -0300, Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? If you work at a lower layer (using such a product, or in similar ways) you have no clear notion of calls. You'll have to analyze the dump later on and actually make use of it. If you work wityh asterisk (search for back to back settings) Asterisk acts as a proxy: it knows about the calls and reconnects them. Thus asterisk can provide you with CDR information, better control of what to record. e.g: you might want to make sure that the system does not record the extra calls it that are sometimes made to your home ;-) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stop log/debug messages into /var/log/messages
Adrian Marsh wrote: When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. hmmm. interesting. need to investigate syslog now. Even me thinks, as far as I've read(abt logger and the existing configuration), it shouldn't be writing to any syslogs. btw, am accessing the * console via ssh. thanks for ur help. - Benjamin Jacob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 12:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Here it is : SIP01*CLI logger show channels Channel Type StatusConfiguration --- --- Console Enabled- Notice Error EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make call from asterisk?
Hi Devraj, May i have your extension.conf working sample?? Thanks you very much. Date: Wed, 5 Sep 2007 16:14:36 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to make call from asterisk? Hi Neoh, All you have to do is configure your VoIp provider as another SIP extension on your Asterisk server and then use extensions.conf to set dialout rules, so when you do dial a number your asterisk server forwards it to the VoIp provider. Examples of extensions.conf can be found at http://www.asteriskguru.com/tutorials/extensions_conf.html http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf I can send you my extensions.conf if you want a working example. I do something very similar with a VoIP provider that provides an SIP interface. Hope this helps. On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, Thanks for your reply.. I am intend to dial using a VOIP provider.(developed by us) Software: x-Lite (SIP softphone) Registration of account number is fine, but for the case when i dial a number, it prompt out a message that the number not found. From my understanding, asterisk can be SIP server? or we need to implement a SIP server to integrate with Asterisk in order to provide full picture of VOIP system? Thanks. Date: Wed, 5 Sep 2007 13:30:21 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to make call from asterisk? Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Live Search: Better results, fast Try it now! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get the new Windows Live Messenger! http://get.live.com/messenger/overview___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P (TDM22P) and aux power.
I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Yes, you only need to connect a power supply if you have FXS boards. Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. I don't remember if it is also needed to provide the potential for the line as well, but I cat testify to the fact that you can comfortably run a TDM400P with 4 FXO boards on it and nothing plugged into the PSU header. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension
On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi Atis, Is your code open source, or are you willing to share your PHP code snippets with me? And thanks for the information on Asterisk's stability. Do you think there is an issue in the implementation or just network/traffic issues? Thanks for your time. Hi, Sorry, but i can't share - it's company's property, and you wouldn't want it, because it includes a bunch of other things - our own libraries, customer recognition, etc, etc.. However, for your purpose - the code for such program would be trivial. All you need is Stomp library for php, and then just convert all data to some format that your program will recognize (i use XML). Also, you might take a look on AJAM - http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29 Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ping
- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Hi to all I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf... Now you suggest to use asterisk realtime (res_config_ldap) or astirectory?? Can I use one of them with version 1.4? thx On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote: No probs. On 29/08/2007, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Thank you once again. Will have to talk it over with my prof before upgrading to Asterisk 1.4. The productive system is currently running on 1.2.6. Thanks Abhishek On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gz from the link http://bugs.digium.com/view.php?id=5768 If you use an old version of res_config_ldap with Asterisk version 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you seek any help via the lists or bug tracker. If you can use the latest release of Asterisk, you should. Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist:
Re: [asterisk-users] Ping
Mike Hammett wrote: Pong -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
I've been trying to send messages to the list for the past 24 hours, but they just aren't going through. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 7:23 AM Subject: [asterisk-users] Ping - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Quoting Doug Lytle ([EMAIL PROTECTED]): Pong The list seems to act weird. I mailed to the list earlier, the message was accepted, but does not appear on the archives nor did i get a bounce or my own listmail back. Though i do see other people posting :/ -- | Only those who will risk going too far, can possibly find out how far you can go. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP...
Carlos Chavez wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. I have a similar setup where I work. I used a Viking PI-1 unit connected to the amp and a SPA-3000 connected to the viking. This gave me overhead paging and ringing. It did require a little tweaking on the SPA side because the PI-1 only provides 12V instead of the normal 48V to CO port. It's been working fine since it was put in about 8 months ago. I think the Viking unit was about $120. More info in it here: http://www.vikingelectronics.com/products/view_product.php?pid=199# -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Agreed. This conversation is working just fine, but the important messages I'm trying to get to go through aren't. I've never had consistent success from posting to asterisk-users. Asterisk-biz seems to work all of the time. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sander Smeenk [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 7:45 AM Subject: Re: [asterisk-users] Ping Quoting Doug Lytle ([EMAIL PROTECTED]): Pong The list seems to act weird. I mailed to the list earlier, the message was accepted, but does not appear on the archives nor did i get a bounce or my own listmail back. Though i do see other people posting :/ -- | Only those who will risk going too far, can possibly find out how far you can go. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Sander Smeenk wrote: Quoting Doug Lytle ([EMAIL PROTECTED]): Pong The list seems to act weird. I mailed to the list earlier, the message was accepted, but does not appear on the archives nor did i get a bounce or my own listmail back. Though i do see other people posting :/ Same thing happened to me a while back. I sent a new message asking a question ..twice.. and neither made it through. However replies to other peoples messages went through just fine. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
Hi i generate a call from the dialplan in this mode: exten = 1002,1,Answer() exten = 1002,2,Dial(SIP/[EMAIL PROTECTED]) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI -- Executing Answer(SIP/host1-0819d0d0, ) in new stack -- Executing Dial(SIP/host1-0819d0d0, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/host-081a2610 is ringing -- SIP/host-081a2610 answered SIP/host1-0819d0d0 -- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610 == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0' i've enabled sip debug, but nothing interesing has been showed host1 is an SJphone and host is a software that implements SIP protocol. Can you help me to guess where is the problem? if i try to create a call from SJphone 2 SJphone all works fine. Is possible that exists a problem in asterisk ? where ? how can i find it ? thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP
I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. At one customer site I use a linksys ATA setup with auto answer, connect the tip and ring of the FXS port to the audio input of the overhead paging system, works fine. Also at another customer site, the paging guys installed a audio input device that activates when ring voltage hits the line, http://www.valcom.com/v-9970.htm . Installed an ATA, nothing special, call that extension and device brings the ATA off-hook then bam, your passing audio to the paging system. You can pick these up for ~$200. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
ACK Mike Hammett wrote: - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP
The phones you are using might support it already (and not even need the system) the grandstreams I have do, but I can't speak for any others. Quoting JR Richardson [EMAIL PROTECTED]: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. At one customer site I use a linksys ATA setup with auto answer, connect the tip and ring of the FXS port to the audio input of the overhead paging system, works fine. Also at another customer site, the paging guys installed a audio input device that activates when ring voltage hits the line, http://www.valcom.com/v-9970.htm . Installed an ATA, nothing special, call that extension and device brings the ATA off-hook then bam, your passing audio to the paging system. You can pick these up for ~$200. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax() problem - fax signal seems to be ignored
Hello,
my configuration is the following:
a TDM400P board with an fxs and fxo daughter boards on it.
I thus connect a fax to my FXS port, after having verified that this port
was correctly functioning. For this, I had tried before with a simple phone,
and with some basic voicemail exten scripts.
Here is my simple dialplan for my fax reception:
exten = 300,1,Ringing()
exten = 300,n,Answer()
exten = 300,n,Set(FAXFILE=/tmp/test.tif)
exten = 300,n,rxfax(${FAXFILE}||debug)
I then dialed 300 on my fax machine, and expected to be lucky and to obtain
a /tmp/test.tif file after faxing completion.
But instead, I always got such error in the /var/log/asterisk/full log file:
[Sep 5 13:42:21] DEBUG[1272] chan_zap.c: Monitor doohicky got event
Ring/Answered on channel 1
[Sep 5 13:42:21] VERBOSE[1298] logger.c: -- Starting simple switch on
'Zap/1-1'
[Sep 5 13:42:22] DEBUG[1298] chan_zap.c: DTMF digit: 3 on Zap/1-1
[Sep 5 13:42:23] DEBUG[1298] chan_zap.c: DTMF digit: 0 on Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: DTMF digit: 0 on Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] devicestate.c: Notification of state change to
be queued on device/channel Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: Enabled echo cancellation on
channel 1
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Ringing'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [EMAIL PROTECTED]:1]
Ringing(Zap/1-1, ) in new stack
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: Requested indication 3 on channel
Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] devicestate.c: Notification of state change to
be queued on device/channel Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Answer'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [EMAIL PROTECTED]:2]
Answer(Zap/1-1, ) in new stack
[Sep 5 13:42:24] DEBUG[1298] devicestate.c: Notification of state change to
be queued on device/channel Zap/1-1
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: Took Zap/1-1 off hook
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Set'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [EMAIL PROTECTED]:3]
Set(Zap/1-1, FAXFILE=/tmp/test.tif) in new stack
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'RxFAX'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [ [EMAIL
PROTECTED]:4]
RxFAX(Zap/1-1, /tmp/test.tif||debug) in new stack
[Sep 5 13:42:24] DEBUG[1298] channel.c: Set channel Zap/1-1 to read format
slin
[Sep 5 13:42:24] DEBUG[1298] channel.c: Set channel Zap/1-1 to write format
slin
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: No provider found, checking
channel drivers for Zap - 1
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: Changing state for Zap/1 -
state 2 (In use)
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: No provider found, checking
channel drivers for Zap - 1
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: Changing state for Zap/1 -
state 2 (In use)
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: No provider found, checking
channel drivers for Zap - 1
[Sep 5 13:42:24] DEBUG[1270] devicestate.c: Changing state for Zap/1 -
state 2 (In use)
[Sep 5 13:42:24] DEBUG[1299] app_queue.c: Device 'Zap/1' changed to state
'2' (In use) but we don't care because they're not a member of any queue.
[Sep 5 13:42:24] DEBUG[1300] app_queue.c: Device 'Zap/1' changed to state
'2' (In use) but we don't care because they're not a member of any queue.
[Sep 5 13:42:24] DEBUG[1301] app_queue.c: Device 'Zap/1' changed to state
'2' (In use) but we don't care because they're not a member of any queue.
[Sep 5 13:43:09] DEBUG[1298] chan_zap.c: Exception on 8, channel 1
[Sep 5 13:43:09] DEBUG[1298] chan_zap.c: Got event On hook(1) on channel 1
(index 0)
[Sep 5 13:43:09] DEBUG[1298] chan_zap.c: disabled echo cancellation on
channel 1
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: Got hangup
[Sep 5 13:43:09] DEBUG[1298] channel.c: Set channel Zap/1-1 to read format
ulaw
[Sep 5 13:43:09] DEBUG[1298] channel.c: Set channel Zap/1-1 to write format
ulaw
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c:
==
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: Fax receive not successful -
result (51) The call dropped prematurely.
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c:
==
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: FLOW FAX Set rx type 13
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: FLOW FAX FAX exchange complete
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: FLOW FAX Set tx type 13
[Sep 5 13:43:09] DEBUG[1298] app_rxfax.c: FLOW FAX FAX exchange complete
[Sep 5 13:43:09] DEBUG[1298] pbx.c: Extension 300, priority 4 returned
normally even though call was hung up
[Sep 5 13:43:09] DEBUG[1298] channel.c: Soft-Hanging up channel 'Zap/1-1'
[Sep 5 13:43:09] DEBUG[1298] channel.c : Hanging up channel 'Zap/1-1'
[Sep 5 13:43:09] DEBUG[1298] chan_zap.c: zt_hangup(Zap/1-1)
[Sep 5 13:43:09] DEBUG[1298] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 8, callwait = -1, thirdcall = -1
[Sep 5 13:43:09]
Re: [asterisk-users] Ping
Dave Fullerton wrote: Same thing happened to me a while back. I sent a new message asking a question ..twice.. and neither made it through. However replies to other peoples messages went through just fine. This may not be the problem, but I've seen this on my NEW post a few times and it was always my fault. My default email is NOT the email I have subscribed to this list. Only subscribers can post. Others don't seem to bounce (why bounce to a spammer) and they are just dropped. However, when I reply to a post, it uses the correct address automatically because the original email originated from the list (with my subscribed address). Make sure your NEW posts are sent from the subscribed address... Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Dial tone came from fxs modules
Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. Best Regards; Wassim _ Windows Live Spaces is here! It’s easy to create your own personal Web site. http://spaces.live.com/signup.aspx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
*nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Bill Andersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 9:04 AM Subject: Re: [asterisk-users] Ping Dave Fullerton wrote: Same thing happened to me a while back. I sent a new message asking a question ..twice.. and neither made it through. However replies to other peoples messages went through just fine. This may not be the problem, but I've seen this on my NEW post a few times and it was always my fault. My default email is NOT the email I have subscribed to this list. Only subscribers can post. Others don't seem to bounce (why bounce to a spammer) and they are just dropped. However, when I reply to a post, it uses the correct address automatically because the original email originated from the list (with my subscribed address). Make sure your NEW posts are sent from the subscribed address... Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. do you have the power cable attached to it. that's what you need to generate a dialtone. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
On Wed, 2007-09-05 at 09:11 -0500, Mike Hammett wrote: *nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. I'm working with Digium's IT department to try to track down the problem. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Once you've loaded the wctdm kernel module, you should get battery on the line but no dialtone. The dialtone isn't put on the line until Asterisk has been configured correctly (see zapata.conf) and restarted. If you're having problems configuring zapata.conf, let us know and we can try to walk you through it. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
and I appreciate it much. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 9:25 AM Subject: Re: [asterisk-users] Ping On Wed, 2007-09-05 at 09:11 -0500, Mike Hammett wrote: *nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. I'm working with Digium's IT department to try to track down the problem. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Died message
You are using safe_asterisk, it will restart automatically Asterisk after it crashes. Original Message Subject: [asterisk-users] Asterisk Died message From: Nitesh Divecha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 04/09/2007 11:23 a.m. Hello All, Anyone knows what does this error message means and where to check for the cause and why it happened? Asterisk on hyperion exited on signal 11. Might want to take a peek. But when I check Asterisk, its running fine... Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. joe a. Correct, it is to provide the ringing voltage on the FXS modules. For systems without internal molex connectors available, there is another option. Digium has created an externally powered supply that can be used with these cards. http://www.digium.com/en/products/hardware/analogpwr.php -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 The switchtype and signalling im using is national, pri_cpe I'm attaching the avaya config details for more information. Please help me to sorted out this problem. - Thanks Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520036 Mobile - +9466596 SIGNALING GROUP Group Number: 1 Group Type: isdn-pri Associated Signaling? y Max number of NCA TSC: 0 Primary D-Channel: 01B0216 Max number of CA TSC: 0 Trunk Group for NCA TSC: Trunk Group for Channel Selection: X-Mobility/Wireless Type: NONE Supplementary Service Protocol: a DS1 CIRCUIT PACK Location: 01B02 Name: ZTE 1 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw CRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? y Near-end CSU Type: other Echo Cancellation? n TRUNK GROUP Group Number: 1Group Type: isdn CDR Reports: y Group Name: OUTSIDE CALLCOR: 14 TN: 1TAC: 801 Direction: two-wayOutgoing Display? n Carrier Medium: PRI/BRI Dial Access? yBusy Threshold: 99Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? nTestCall ITC: rest Far End Test Line No: TestCall BCC: 4 TRUNK PARAMETERS Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical Digital Loss Group: 13 Calling Number - Delete: Insert: Numbering Format: Bit Rate: 1200 Synchronization: sync Duplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 TRUNK FEATURES ACA Assignment? nMeasured: none Wideband Support? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: n Send Calling Number: y Used for DCS? n Suppress # Outpulsing? nNumbering Format: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Send Connected Number: y Send UUI IE? y Send UCID? n Send Codeset 6/7 LAI IE? y Ds1 Echo Cancellation? n US NI Delayed Calling Name Update? n SBS? n Network (Japan) Needs Connect Before Disconnect? n DS1 CIRCUIT PACK Location: 01B01 Name: ZTE 4 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw CRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? y Near-end CSU Type: other Echo Cancellation? n ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Died message
It will not after some types of crashes. /b On Sep 5, 2007, at 9:43 AM, Perssy Llamosas wrote: You are using safe_asterisk, it will restart automatically Asterisk after it crashes. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presentation and mISDN
Hi guys,
is it possible to set caller presentation with mISDN? I tryie with
SetCallerPres() and CallingPres without success...
exten = s,1,ChanIsAvail(mISDN/1)
exten = s,2,CallingPres(32)
exten = s,3,Set(CALLERID(num)=e.164_number)
exten = s,4,Dial(${CUT(AVAILCHAN||1)}/${ARG2})
Anyone can help me ?
Thanks in advance
Giordano
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.485 / Virus Database: 269.13.5/990 - Release Date: 04/09/2007 22.36
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[asterisk-users] Dialplan regexp
Hi,
Can anyone tell me why the below dialplan doesn't filter off dialed
numbers for 01793520158, and jump to local,priority1
If I change it to :
exten = 01793520158,1,Goto(local,${EXTEN:-3},1)
then it works fine (but that's too specific)...
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
Adrian Marsh
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Re: [asterisk-users] Asterisk Died message
On Wed, Sep 05, 2007 at 09:43:20AM -0500, Perssy Llamosas wrote: You are using safe_asterisk, it will restart automatically Asterisk after it crashes. Or will contantly die, clog the logs and make debugging the problem more difficult than it is. Or you might have two safe_asterisk processes trying to restart asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPBroker vs SIPgate
I had to turn Sipbroker off at one point, as I found that some Conf. Calls on a 3rd party system didn't like the DTMF being passed (users unable to enter conferences). I traced all the failures to calls passing out via SIPbroker, disabled it so the calls went via PSTN and all was well.. Now I'm trying to re-work the call logic to include it again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SIP Sent: 04 September 2007 18:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIPBroker vs SIPgate Seriously, from our experience, SIPBroker IS the best way to interact with all the open networks. For any closed networks, you might create special rules for interaction, but that would rely on setting up a deal with the respective destination network to actually ALLOW your calls. There are some pay per play networks that do peering automagically (such as XConnect), but it's a cost per connected call (granted, a tiny one, but still a cost), and it won't guarantee you any better connectivity to a closed network than, say, SIPBroker. N. Adrian Marsh wrote: Yeah, I can see that now after testing it all - but this is what raised my question.. What IS the best mechanism for all the VoIP servers/networks to interact ? Setting up individual agreements for each network is so 1980's, and in this modern world there must be a better solution.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SIP Sent: 04 September 2007 15:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIPBroker vs SIPgate Adrian Marsh wrote: All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is we don't support SIPBroker... So whats the easiest way to support SIP SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users SIP - SIP calling across networks really only works if the receiving network allows incoming calls from non-local networks. SIPgate does not, so unless you're registered on the SIPgate network, calling another SIPgate user from your SIPgate number, it won't accept the call. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
On Wed, Sep 05, 2007 at 08:26:25PM +0530, Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 Maybe try instead: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 The switchtype and signalling im using is national, pri_cpe national? Is this the one that should be used there? Not euroisdn? (Though this is unrelated to the current issue. we're not in chan_zap yet, so zapata.conf is unrelated at the moment. It will be once you get over the above error) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
On Wed, Sep 05, 2007 at 10:27:48AM -0400, Jared Smith wrote: On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Once you've loaded the wctdm kernel module, you should get battery on the line but no dialtone. The dialtone isn't put on the line until Asterisk has been configured correctly (see zapata.conf) and restarted. If you're having problems configuring zapata.conf, let us know and we can try to walk you through it. Right. Just a small note: If you really want to get a nice dialtone without asterisk, use the testing usility fxstest. 'make fxstest' to build it. ./fxstest to run it. If you used a more recent version of Zaptel, you had a man page for it -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Died message
On 9/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: Or you might have two safe_asterisk processes trying to restart asterisk. A symptom of this (when Asterisk is not actively crashing) is constant remote UNIX connection messages on the console every few seconds (assuming you have nothing that legitimately polls Asterisk using 'asterisk -rx' running). The solution is to use ps to find out which of the safe_asterisk processes owns the actual running copy of Asterisk (using pid and ppid) and then kill the other one. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A102d sangoma's card and ztdummy
Sin you have sangoma card , it will act as timer . You need to install meetme ( app_conference is not very stable last time i read ) . On 01/09/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. -- Yahoo! oneSearch: Finally, mobile search that gives answershttp://us.rd.yahoo.com/evt=48252/*http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC, not web links. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan regexp
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me why the below dialplan doesn't filter off dialed
numbers for 01793520158, and jump to local,priority1
If I change it to :
exten = 01793520158,1,Goto(local,${EXTEN:-3},1)
then it works fine (but that's too specific)...
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
I'm not sure about [56][0-9], but..
_0. will be executed before 017... because it is first in ASCII
sorting. If you need 017 before, you should change _0. to _0X.
CLI show dialplan is your friend ;-)
Regards,
Atis
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IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
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[asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3.1 released
Asterisk-Java 0.3.1, a free Java library for Asterisk PBX integration, has been released. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. Asterisk-Java 0.3.1 is a maintenance release that solves the following issues: * [AJ-81] - executeCliCommand() always executes show voicemail users * [AJ-86] - getChannelByName doesn't return the latest channel * [AJ-80] - getMeetMeRooms() should only return active rooms * [AJ-68] - Support for Bridge Action * [AJ-74] - Support Strategy property in QueueParamsEvent Asterisk-Java takes advantage of the features of Java 5.0 and therfore requires a Java Virtual Machine of at least version 1.5.0. Asterisk-Java is used in several commercial environments and by the following Open Source projects: * Asterisk-JTAPI JTAPI implementation for Asterisk. http://asterisk-jtapi.sf.net/ * Asterisk-IM A plugin for the Openfire XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients. http://www.igniterealtime.org/projects/openfire/ * Asterisk Desktop Manager (ADM) A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more. http://adm.hamnett.org/ Asterisk-Java is available under Apache 2.0 license at http://asterisk-java.org signature.asc Description: OpenPGP digital signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Relay Problems
Hi, I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Regards, Joseph ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + LDAP or RADIUS
Hi to all, I've installed Asterisk 1.4 and all function very well. Now I need to use LDAP or RADIUS instead of sip.conf since all the trusted users have an account on LDAP/RADIUS. Any suggestions...try astirectory (but is for asterisk 1.2.x, I've 1.4.9) or Asterisk realtime LDAP (it is only for 1.2 or not) Thx for all! bye -- Alessandro R. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan regexp
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Sent: 05 September 2007 17:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan regexp
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me why the below dialplan doesn't filter off dialed
numbers for 01793520158, and jump to local,priority1
If I change it to :
exten = 01793520158,1,Goto(local,${EXTEN:-3},1)
then it works fine (but that's too specific)...
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
I'm not sure about [56][0-9], but..
_0. will be executed before 017... because it is first in ASCII
sorting. If you need 017 before, you should change _0. to _0X.
CLI show dialplan is your friend ;-)
Regards,
Atis
--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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[asterisk-users] special kind of billing
Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Cheers, Kate ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 Remove dchan=16 from zaptel.conf. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan regexp
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
You can also impose a finer level of control over the order extensions are
searched in by putting them in different contexts and using include to
pull them in in a specific order:
[foo]
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
include = bar
[bar]
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
Dialing 01793520158 would match the longer pattern in this case. The search
is done in the initial context, then in each included context in the order
they were included.
There's more info here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
--
j.
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Re: [asterisk-users] Asterisk + LDAP or RADIUS
RADIUS does two things 1) authentication 2) accounting (well, actually, 3 things, but I see no difference of authorising and authentication) accounting is easy for asterisk-1.4, there're CDR (call detail record) which stores call in radius out of box. as for authentication/authorising against RADIUS/LDAP, there's no simple solution yet for asterisk. On 9/5/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all, I've installed Asterisk 1.4 and all function very well. Now I need to use LDAP or RADIUS instead of sip.conf since all the trusted users have an account on LDAP/RADIUS. Any suggestions...try astirectory (but is for asterisk 1.2.x, I've 1.4.9) or Asterisk realtime LDAP (it is only for 1.2 or not) Thx for all! bye -- Alessandro R. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
Thomas Kenyon wrote: Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Yes, you only need to connect a power supply if you have FXS boards. Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. I don't remember if it is also needed to provide the potential for the line as well, but I cat testify to the fact that you can comfortably run a TDM400P with 4 FXO boards on it and nothing plugged into the PSU header. That is correct. You *only* need the power connector plugged in for FXS modules. FXO modules do not need them. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 It's fairly self explanatory. CAS stands for Channel Associated Signalling. That means signalling is passed on the same channel that the media is, like in robbed bit signalling protocols like FXO, FXS, EM, etc. Since you are using a PRI which does not contain inband signalling, but rather out of band signalling, you need to set it to `ccs` instead of `cas` (in your span= line) which stands for Common Channel Signalling. This is for signalling modes such as PRI or SS7 which use a dedicated channel to do call related signalling. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
Just to be clear, I thought that dialtone provision didn't require the power cable, just generating ring voltages? Can anyone say? Moj Anthony Messina wrote: On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. do you have the power cable attached to it. that's what you need to generate a dialtone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can asterisk give half-ring periodically for MWI?
Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Hi, There isn't an astirectory driver for Asterisk version 1.4. So I guess you'll have to use the asterisk realtime (res_config_ldap) driver. cheers Abhishek On 9/5/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf... Now you suggest to use asterisk realtime (res_config_ldap) or astirectory?? Can I use one of them with version 1.4? thx On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote: No probs. On 29/08/2007, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Thank you once again. Will have to talk it over with my prof before upgrading to Asterisk 1.4. The productive system is currently running on 1.2.6. Thanks Abhishek On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gz from the link http://bugs.digium.com/view.php?id=5768 If you use an old version of res_config_ldap with Asterisk version 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you seek any help via the lists or bug tracker. If you can use the latest release of Asterisk, you should. Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver
Re: [asterisk-users] DTMF Relay Problems
In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Try the patch at http://bugs.digium.com/view.php?id=10535 and see if it helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Benchmark
Hi all, Please mention your real life experience with Asterisk about how many concurrent calls a single server has handled for you. Please don't tell me it depends on the Hardware or ., I want your experiences. Someone might used it as a calling card with a2billing on a single box with 60 concurrent connection, so please mention what was your Hardware and what was the applications and how many simultaneous calls (with codec conversion or without it please mention) 1- How many Calls? 2- Hardware? 3- Applications. 4- off course other factors can be added to this list, but it is upto you to mention it or not. thank you all, AFShin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Quoting Jared Smith ([EMAIL PROTECTED]): *nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. I'm working with Digium's IT department to try to track down the problem. As it may help you follow the message i 'lost' through your systems: It had Message-ID header [EMAIL PROTECTED] Your server lists.digium.com.s8a1.psmtp.com [64.18.7.10] accepted it with a TLS connection but did not reply with a queue message-id. HTH, Sander. -- | If you look like your passport picture, you probably need the trip. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unsuscribe
Denied. On 9/4/07, Moshe at Talk'n'Save [EMAIL PROTECTED] wrote: please unsubscribe Moshe Wahrhaftig IT Manager Talk'n'Save Israel: 02-655-0313 Cell: 052-2771738 USA: 516-204- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Rodriguez Sent: Monday, September 03, 2007 10:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Rechazo de llamada en triangulacion deasterisk. Gracias Alex. Lo pondré el Lunes y te tire como van las cosas. Guillermo El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
The SIPuras support it, Asterisk analog does not, as far as I know. Matt wrote: The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
Hi, thanks for the reply. This capability is provided by Sipura ATAs (apparently they do it each time they process SIP REGISTER messages with MWI). The periodic ring works when the same analog phone is connected the Sipura ATA. But not when it is connected to the TDM400p. So to reiterate, what I'm looking for is a way to get the half-ring generated by asterisk and/or TDM400p, WITHOUT the use of a SIP-based ATA. - Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:40:08 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
Do Linksys PAP2Ts support it and if so, where is the setting? On 9/5/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The SIPuras support it, Asterisk analog does not, as far as I know. Matt wrote: The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
On 9/5/07, Ricardo Gemignani [EMAIL PROTECTED] wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You need double the number of E1 ports in your Asterisk machine as there are E1 ports in the PBX. So say the PBX has 3 E1 ports, you need 6 on your Asterisk machine. Basically you are going to put Asterisk to interface with the PSTN on half the ports, the otehr half will continue to be the same to the PBX. You use cpe_net on PBX end of things in Asterisk,. so the PBX does not need to be re-configured. There is no need for tapping the Asterisk will just be in the middle of the PBX and PSTN. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special kind of billing
On 9/5/07, Kate Kretz [EMAIL PROTECTED] wrote: Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Honestly I've looked and I've looked and I've looked. There IS NO out-of-the-box, compatible with Asterisk solution that does any sort of billing provisioning for a reasonable price. None. However you can find 90% of the functions you need -- call rating least cost routing as free open source and have your programmer write the glue you need. Take a look at the voip-info.org site and also google.com. You'll probably also get some emails from a few people offering you solutions starting around 20K. The Didx people sell their billing solution for something like USD 50K, if not more I dont recall the exact number but it was way overpriced they can barely write a user friendly UI honestly. If you are willing to spend this much, evaluate with great care and make sure you get a written warranty which explicitly spells out the feature set you are expecting! DO NOT purchase anything from AgileCO or AgileBill they are a scam we paid them almost $10,000 and their product is crap and NOONE will answer your phone calls, much less email messages. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax() problem - fax signal seems to be ignored
On 9/5/07, Pirlouwi [EMAIL PROTECTED] wrote:
Hello,
my configuration is the following:
a TDM400P board with an fxs and fxo daughter boards on it.
I thus connect a fax to my FXS port, after having verified that this port
was correctly functioning. For this, I had tried before with a simple phone,
and with some basic voicemail exten scripts.
Here is my simple dialplan for my fax reception:
exten = 300,1,Ringing()
exten = 300,n,Answer()
exten = 300,n,Set(FAXFILE=/tmp/test.tif)
exten = 300,n,rxfax(${FAXFILE}||debug)
Why? exten = 300,1,rxfax(/tmp/test.tif||debug) would do the same
exact thing. No need to indicate ringing and no need to answer the
call. Besides that it is just incorrect you are never going to have
correct answer supervision on an analog line, so don't even try.
I then dialed 300 on my fax machine, and expected to be lucky and to obtain
a /tmp/test.tif file after faxing completion.
But instead, I always got such error in the /var/log/asterisk/full log file:
What if you just use a regular analog phone and dial 300? What
happens? What if you remove the ||Debug from your RxFax dialstring?
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Ringing'
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: Took Zap/1-1 off hook
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Set'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [EMAIL
PROTECTED]:3]
Set(Zap/1-1, FAXFILE=/tmp/test.tif) in new stack
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'RxFAX'
Notice how your own logs prove that 0ms elapse between the time you
incorrectly indicate ringing on the channel and the time RxFax begins.
I have enabled the #define LOG_FAX_AUDIO inside spandsp library, and two
audio files (fax-rx-audio-b7933500-070905134224 and
fax-tx-audio-b7933500-070905134224) appeared in /tmp.
Just listen into the line. When you execute RxFax it will play fax
tones just as if another faxmachine answered -- not CNG tones
This is not the case in my setup. What did I wrong?
Thx for your help.
What version of Linux, Asterisk, Zaptel, SpanDSP app_rxfax are you using?
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Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
On 9/5/07, Matt [EMAIL PROTECTED] wrote: Do Linksys PAP2Ts support it and if so, where is the setting? I don't know about PAP2T but SPA2102 does. Basically anything that is similar to the Sipira-SPA firmware, I don't know how familar you are with them but if your webinterface looks like this: http://www.3cx.com/voip-gateways/images/sipura1.jpg 1) the adapter is based on the original Sipura SPA designs firmwares 2) you should have the option. Honestly I think the PAP2T is one that is based on totally Linksys design. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having troublespeakingclearly
Bingo! That was it. Well, it's got it to 98% there. I can play with it now and tweek it. Todd - Original Message - From: Kai-Uwe Jensen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 12:36 AM Subject: Re: [asterisk-users] Cepstral's Allison is having troublespeakingclearly How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. On 9/3/07, Todd Reese [EMAIL PROTECTED] wrote: OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there. TIA, Todd -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan regexp
On Wed, 2007-09-05 at 12:57 -0400, James FitzGibbon wrote:
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked
quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
You can also impose a finer level of control over the order extensions
are searched in by putting them in different contexts and using
include to pull them in in a specific order:
[foo]
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
include = bar
[bar]
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
Dialing 01793520158 would match the longer pattern in this case. The
search is done in the initial context, then in each included context
in the order they were included.
There's more info here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config
+extensions.conf+sorting
James speaks the truth. Within a single context, the algorithm tries to match
EVERY POSSIBLE extension. The one that scores the best wins. The more specific
the pattern, the higher the score. So, _0. would lose to _017. if they
both
matched.
If ANY pattern matches, the include path will not be followed.
murf
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] special kind of billing
On Wed, 2007-09-05 at 22:44 +0600, Kate Kretz wrote: Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Yes, We are using a2billing [1]. You can define serveral trunks and add rates for the destinations, the a2billing can use low cost routing and gives to you a detailed call detail record with the ammount of sell, buy, profit, margin and markup. You can learn to use with this small guide (spanish): http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos [1] www.asterisk2billing.org Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
Date: Wed, 5 Sep 2007 09:21:19 -0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Dial tone came from fxs modules Just to be clear, I thought that dialtone provision didn't require the power cable, just generating ring voltages? Can anyone say? Moj Anthony Messina wrote: On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. do you have the power cable attached to it. that's what you need to generate a dialtone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi: I checked the power cable and its plugged in the TDM, Is there anything else to check? _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
On 9/5/2007 at 1:06 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Yes, you only need to connect a power supply if you have FXS boards. Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. I don't remember if it is also needed to provide the potential for the line as well, but I cat testify to the fact that you can comfortably run a TDM400P with 4 FXO boards on it and nothing plugged into the PSU header. That is correct. You *only* need the power connector plugged in for FXS modules. FXO modules do not need them. Thanks to all who responded. My hunt for cheap, err, inexpensive, Polycom's continues. joe a. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
Yeah, it's a hack for half-rings, but a little less so for putting someone right into voicemailmain without delay. Moj Justin Ridge wrote: That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
Agreed. I appreciate your suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 5:55:27 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? Yeah, it's a hack for half-rings, but a little less so for putting someone right into voicemailmain without delay. Moj Justin Ridge wrote: That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 14. Re: ztcfg error : TE110p error with CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 Remove dchan=16 from zaptel.conf. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
The Polycom hardphones do it by defaultBUT a colleague of mine worked in a large office and she said that monday morning people would be driven mad by almost every phone on the floor making that beeble-bup noise...over and over and over PaulH On Wed, 2007-09-05 at 10:32 -0700, Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx XML Apps (was: Re: Cisco Directory Format)
On 9/4/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. Cisco has a book that covers some of this. Not sure the name. I've got it, but haven't had a chance to do anything yet. On Sat, 2007-09-01 at 12:00 -0500, [EMAIL PROTECTED] wrote: Date: Sat, 1 Sep 2007 12:14:49 -0400 From: Time Bandit [EMAIL PROTECTED] Subject: Re: [asterisk-users] Cisco Directory Format To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 A little off topic (sorry..:) ) but anyone know what format Cisco phones use for their contact dirctories. I want to set up my contact lists on the phone, and cannot seem to get any info on it. I am working with a 7970 on Asterisk 1.4.8. 7940 and 7960 use this format of XML file (probably the same on 7970) CiscoIPPhoneDirectory TitleEmployee directory/Title PromptOpen Source Rock/Prompt DirectoryEntry NameEmployee A/Name Telephone7001/Telephone /DirectoryEntry DirectoryEntry NameEmployee B/Name Telephone7002/Telephone /DirectoryEntry /CiscoIPPhoneDirectory Check also Open 79XX XML Directory : http://web.csma.biz/apps/xml_xmldir.php hope that help -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
Thanks to all who responded. My hunt for cheap, err, inexpensive, Polycom's continues. How cheap? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] remove unnecessary text (was: Re: Can asterisk give half-ring periodically for MWI?)
Let me quote oej: Make sure that you remove unnecessary text when you reply I don't need messages to tell me *5* times about Astricon, who provides the bandwidth and how to unsubscribe. I'm sure this has been posted a dozen times but please http://learn.to/quote Thanks, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove unnecessary text (was: Re: Can asterisk give half-ring periodically for MWI?)
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote: I don't need messages to tell me *5* times about Astricon, who provides the bandwidth and how to unsubscribe. You sure about that unsubscribe part? People do seem to miss it :P /b ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with calling queues
You need to log your agents in - or set your queue members to be SIP accounts. (which is probably the best solution) PaulH On Wed, 2007-09-05 at 16:53 +1000, Joshua Small wrote: Hi, I’ve just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date TimeZone: Thu Sep 6 02:37:11 EST 2007 I’ve used the Asterisk GUI for setup with two IP handsets, one VOIP account with a telco and one PSTN. The server correctly allows: - Handsets to call each other - Calls outbound through both PSTN or VOIP I’m having an issue with incoming calls however. If I configure “incoming calls” coming over my PSTN to a single user, it works correctly (that handset rings, can pickup etc). However if I define a call queue which consists of both these handsets, neither ever rings. Looking at the console, I see this: -- Started music on hold, class 'default', on Zap/1-1 [Sep 6 02:22:51] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: Unexpected control subclass '2' [Sep 6 02:22:54] WARNING[5955]: channel.c:2129 ast_waitfordigit_full: Unexpected control subclass '2' The error repeats until the caller hangs up. I’ve posted all the config that I felt was relevant here, let me know if you need more. This was all written by Asterisk-GUI. I realise there’s a lot more configuration but given that things work fine when I set the receive to a single agent, I assumed it was a queue issue. Users.conf [6001] callwaiting = yes context = numberplan-custom-1 email = [EMAIL PROTECTED] fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = SECRET threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 Queues.conf [6003] fullname = All of us strategy = ringall timeout = wrapuptime = autofill = yes autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = member = Agent/6001 member = Agent/6002 extensions.conf - broken [DID_trunk_2] include = default exten = _X.,1,Goto(default|6003|1) exten = s,1,Goto(default|6003|1) extensions.conf – works but only sends to a single handset [DID_trunk_2] include = default exten = _X.,1,Goto(default|6001|1) exten = s,1,Goto(default|6001|1) Any assistance appreciated. Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 | www.visinet.com.au This e-mail is intended for use by the named recipients only and contains confidential information. Opinions and other information in this message that pertain to the sender's employer and its products and services represent the opinion of the sender and not necessarily those of the employer. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove unnecessary text
Brian West wrote: On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote: I don't need messages to tell me *5* times about Astricon, who provides the bandwidth and how to unsubscribe. You sure about that unsubscribe part? People do seem to miss it :P Good point. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Ignoring SIP ACK's on 487 Responses
Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The retransmits do not occur on a 200 Ok Response. When the INVITE is cancelled the CANCEL request is acted on correctly and the cll is cancelled and the only problem is the 6 retransmits of the INVITE response everytime a call is cancelled. I've confirmed that the ACK request is getting through to the Asterisk 1.4 server and also checked that all the required transaction fields in the ACK are correct (Call-Id, From and Via branch of original INVITE and To of response). I've also checked with two different user agents (Bria softphone and Polycom IP300) and both exhibit the same problem. Below is a trace of the relevant SIP messages. I'm wondering if Asterisk is not coping with the Contact header not being present although no message is coming up on the console to that effect. Regards, Greyman. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKbfe545ae83ef7d0ec9fe44ea063d72c67f4bc926 Via: SIP/2.0/UDP 192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543- To: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=6e2df459 Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI. CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED] Max-Forwards: 69 Record-Route: sip:10.0.0.1;lr User-Agent: Bria release 2.0 stamp 40829 Content-Type: application/sdp Content-Length: 616 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Retransmitting #4 (NAT) to 10.0.0.1: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKbfe545ae83ef7d0ec9fe44ea063d72c67f4bc926;received=194.213.29.100 Via: SIP/2.0/UDP 192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543- From: sip:[EMAIL PROTECTED];tag=6e2df459 To: sip:[EMAIL PROTECTED];tag=as3a3770b5 Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI. CSeq: 2 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- SIP read from 10.0.0.1 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK13da7f631fc65ea06570ce13f322cc7b142074a2 Via: SIP/2.0/UDP 192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543- To: sip:[EMAIL PROTECTED];tag=as3a3770b5 From: sip:[EMAIL PROTECTED];tag=6e2df459 Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI. CSeq: 2 ACK Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
Have you tried asterisk -rvvv? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Wednesday, September 05, 2007 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0 Hi i generate a call from the dialplan in this mode: exten = 1002,1,Answer() exten = 1002,2,Dial(SIP/[EMAIL PROTECTED]) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI -- Executing Answer(SIP/host1-0819d0d0, ) in new stack -- Executing Dial(SIP/host1-0819d0d0, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/host-081a2610 is ringing -- SIP/host-081a2610 answered SIP/host1-0819d0d0 -- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610 == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0' i've enabled sip debug, but nothing interesing has been showed host1 is an SJphone and host is a software that implements SIP protocol. Can you help me to guess where is the problem? if i try to create a call from SJphone 2 SJphone all works fine. Is possible that exists a problem in asterisk ? where ? how can i find it ? thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 14. Re: ztcfg error : TE110p error with CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
On Thu, Sep 06, 2007 at 05:48:57AM +0530, Vidura Senadeera wrote: Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Please leave dchan=16 , and replace 'cas' with 'ccs' in the span= line. (if you were using CAS, you would have a different configuration. You wouldn't have bchans ). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alphabetical extension patterns
Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
I have the same setup asterisk-1.4.11 with TE120P two port E1 card with is connected with avaya system but signaling is Qsig becase i want unified dialplan my configuration /etc/zaptel.conf ### Digium TE120P Card Configuration # # E1 port 1 span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 # E1 port 2 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone = in defaultzone=in /etc/asterisk/zapata.conf group=1 context=from-avaya signalling=pri_net channel = 1-15,17-31 group=2 context=from-pstn signalling=pri_cpe channel = 32-46,48-62 *Notes : - my avaya system clock is slave mode and my asterisk is master thats why i use pri_net on avaya Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 The switchtype and signalling im using is national, pri_cpe I'm attaching the avaya config details for more information. Please help me to sorted out this problem. - Thanks Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520036 Mobile - +9466596 SIGNALING GROUP Group Number: 1 Group Type: isdn-pri Associated Signaling? y Max number of NCA TSC: 0 Primary D-Channel: 01B0216 Max number of CA TSC: 0 Trunk Group for NCA TSC: Trunk Group for Channel Selection: X-Mobility/Wireless Type: NONE Supplementary Service Protocol: a DS1 CIRCUIT PACK Location: 01B02 Name: ZTE 1 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw CRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? y Near-end CSU Type: other Echo Cancellation? n TRUNK GROUP Group Number: 1Group Type: isdn CDR Reports: y Group Name: OUTSIDE CALLCOR: 14 TN: 1TAC: 801 Direction: two-wayOutgoing Display? n Carrier Medium: PRI/BRI Dial Access? yBusy Threshold: 99Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? nTestCall ITC: rest Far End Test Line No: TestCall BCC: 4 TRUNK PARAMETERS Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical Digital Loss Group: 13 Calling Number - Delete: Insert: Numbering Format: Bit Rate: 1200 Synchronization: sync Duplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 TRUNK FEATURES ACA Assignment? nMeasured: none Wideband Support? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: n Send Calling Number: y Used for DCS? n Suppress # Outpulsing? nNumbering Format: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Send Connected Number: y Send UUI IE? y Send UCID? n Send Codeset 6/7 LAI IE? y Ds1 Echo Cancellation? n US NI Delayed Calling Name Update? n SBS? n Network (Japan) Needs Connect Before Disconnect? n DS1 CIRCUIT PACK Location: 01B01 Name: ZTE 4 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw
[asterisk-users] Choppy sound while converting alaw to ulaw
Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw = ulaw is choppy, ulaw = alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
Off-topic to the original thread. I just wonder what you meant in your configuration: On Wed, Sep 05, 2007 at 09:58:19PM -0700, satish patel wrote: I have the same setup asterisk-1.4.11 with TE120P two port E1 card with is connected with avaya system but signaling is Qsig becase i want unified dialplan my configuration /etc/zaptel.conf ### Digium TE120P Card Configuration # # E1 port 1 span=1,1,0,ccs,hdb3 '1' in the timing parameter: You tell Zaptel to take timing from that span. bchan=1-15,17-31 dchan=16 # E1 port 2 span=2,0,0,ccs,hdb3 '0' in the timing parameter. You tell asterisk not to take timing from that span (and hence implicitly provide timing). bchan=32-46,48-62 dchan=47 loadzone = in defaultzone=in /etc/asterisk/zapata.conf And what do you use for switchtype? group=1 context=from-avaya signalling=pri_net channel = 1-15,17-31 Span 1 (you timing master) is the Avaya? group=2 context=from-pstn signalling=pri_cpe channel = 32-46,48-62 Span 2 is the PSTN? *Notes : - my avaya system clock is slave mode and my asterisk is master thats why i use pri_net on avaya So you take timing from the Avaya switch rather than from the PSTN? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
yes, i've tried asterisk -r i've also tried sip debug, but i can't reach any error... only that the cmmunication is finished. On 9/6/07, Shonga_Kerz [EMAIL PROTECTED] wrote: Have you tried asterisk -rvvv? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Wednesday, September 05, 2007 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0 Hi i generate a call from the dialplan in this mode: exten = 1002,1,Answer() exten = 1002,2,Dial(SIP/[EMAIL PROTECTED]) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI -- Executing Answer(SIP/host1-0819d0d0, ) in new stack -- Executing Dial(SIP/host1-0819d0d0, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/host-081a2610 is ringing -- SIP/host-081a2610 answered SIP/host1-0819d0d0 -- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610 == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0' i've enabled sip debug, but nothing interesing has been showed host1 is an SJphone and host is a software that implements SIP protocol. Can you help me to guess where is the problem? if i try to create a call from SJphone 2 SJphone all works fine. Is possible that exists a problem in asterisk ? where ? how can i find it ? thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX machine connect with audiocode SIP device
Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
