Re: [asterisk-users] ZAP device reference in Zaptel 1.4
On Wed, Apr 04, 2007 at 02:55:02PM +1000, Devraj Mukherjee wrote: Also I can cat /dev/zap/3 and /dev/zap/4 and they respond to the various changes in signals You're looking at the kernel level . Maybe it's fine there, but asterisk does not know about it. What is the output of: zap show channels What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? On 4/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 04, 2007 at 02:55:02PM +1000, Devraj Mukherjee wrote: Also I can cat /dev/zap/3 and /dev/zap/4 and they respond to the various changes in signals You're looking at the kernel level . Maybe it's fine there, but asterisk does not know about it. What is the output of: zap show channels What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem
Drew Gibson wrote: This looks like the same issue I have with Astra phones, see the thread Multi-line phones - Asterisk uses wrong callerid. I do not know of a resolution for this yet. I had the same problem, so I've modified the peer matching function to match called number with caller id number besides the usual IP address and port number. The patch isn't pretty, but works for me: http://toxygen.net/~wojtekka/asterisk-1.2.17-sip-cid.patch I'll post it to the bugtracker if anyone else considers it useful. Regards, Wojtek ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
I'm a friend of fb, and he is in vacation until the 11th of March. I will try to unsubscribe him from the digest. Sorry jb john beaman a écrit : Ah, yes. One of the many differences between the US and the rest of the world. [EMAIL PROTECTED] 4/3/2007 2:52:16 PM john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... This is from April 2nd to April 11th. Doug - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jean-Baptiste Bellet Ingénieur Développement Lucyde SAS Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex +33 (0)5 34 31 86 36 http://www.lucyde.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 14
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remastering asterisk
Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: System from AMI
Richard Lyman wrote: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: dummy Exten: 2 Priority: 1 In extensions.conf [dummy] Exten = _X,1,System(*some command*) remember your permissions OK, thank you! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extra field
Hi, I am using my asterisk server like a gateway and one provider ask me to pass an extra field with the IP of the peer that is using the connection, probably to have more control on the authentication. I was wondering how I can implement this. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
Hi Tzafir / List. Thank you for your reply. I have run: make clean Configure Make Make install I get no compile errors, but still the same problems if I try to insmod zaptel As you suggested I tried modinfo zaptel Which resulted in: modinfo: could not find module zaptel I also tried depmod with the same result and finally I tried insmod ./ztdummy from the src/zaptel-1.4.1 directory which resulted in: insmod: error inserting './ztdummy.ko': -1 Invalid module format Your continued help is much appreciated. Chris Original Message Reads. Message: 8 Date: Tue, 3 Apr 2007 19:57:40 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Have you run 'make install'? What is the output of modinfo zaptel Any change if you run: depmod [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory insmod ./ztdummy.ko But it should fail (e.g: because zaptel is not loaded). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com http://www.xorcom.com/ iax:[EMAIL PROTECTED]/tzafrir -- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
make clean configure make linux26 make install perhaps Bails Chris Blunt wrote: Hi Tzafir / List. Thank you for your reply. I have run: make clean Configure Make Make install I get no compile errors, but still the same problems if I try to insmod zaptel As you suggested I tried modinfo zaptel Which resulted in: modinfo: could not find module zaptel I also tried depmod with the same result and finally I tried insmod ./ztdummy from the src/zaptel-1.4.1 directory which resulted in: insmod: error inserting './ztdummy.ko': -1 Invalid module format Your continued help is much appreciated. Chris Original Message Reads. Message: 8 Date: Tue, 3 Apr 2007 19:57:40 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Have you run 'make install'? What is the output of modinfo zaptel Any change if you run: depmod [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory insmod ./ztdummy.ko But it should fail (e.g: because zaptel is not loaded). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
I tested this patch to asterisk-1.2.17 with zaptel-1.2.10 and redhat 9. However, to compile on my environment, 'first' function was replaced by the 'firstword' function. Regards, Xiu -- Xiu YuShen 2007/3/17, Tzafrir Cohen [EMAIL PROTECTED]: On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote: Whats the non-workaround solution? Is there one? http://bugs.digium.com/view.php?id=9303 Please test. Wasn't there an existing issue on this one? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Localise VM_DATE timestamp like the voicemessage envelope
Hello, is there anyway or any plan to have the date/time stamp that's printed in an outgoing voicemail notification email to NOT be the date/time of the (*) machine but infact correspond to the timezone set for the subscriber under the TZ variable? I have the (*) machine set to UTC and when the notification email goes out, it prints out the date/time of the machine at which the voicemail was left but when you hear the envelope of the voicemail, it's the subscriber's local timezone. Which is ofcourse the correct behaviour. But this is not the same for the notification email. So, Is there a smart way of modifying the VM_DATE variable to read the DB to do what the envelope does? Perhaps a real smart DialPlan trick to pick that up during the time the voicemail is being left or something? If I were to use the externnotify, then how would I go about maybe ceating a script that can access the DB, get the subscriber's timezone, convert the machine's UTC time to the subscriber's timezone, and then create the same message? Just wondering if someone has actually solved this already and would like to help before I start to maybe writing a script of my own. many thanks, \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium B410P Need Help
Hi All Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting follow message asterisk shows no port. Would be kind enough if somebody help me. Regards Farooq #misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extra field
Hi, Could you please explain what your provider is expecting? You should only have to provide your public IP address. On 4/4/07, Il Neofita [EMAIL PROTECTED] wrote: Hi, I am using my asterisk server like a gateway and one provider ask me to pass an extra field with the IP of the peer that is using the connection, probably to have more control on the authentication. I was wondering how I can implement this. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
On Wed, Apr 04, 2007 at 06:50:50PM +0900, Xiu YuShen wrote: 2007/3/17, Tzafrir Cohen [EMAIL PROTECTED]: http://bugs.digium.com/view.php?id=9303 Please test. I tested this patch to asterisk-1.2.17 with zaptel-1.2.10 and redhat 9. However, to compile on my environment, 'first' function was replaced by the 'firstword' function. Could you please be more specific? Replace where, exactly? In asterisk? In zaptel? In which source file? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what the cable to connect with digium TE110 and avaya s8300
Hi all, what kind of cable to connect TE110 and avaya between Straight cable or crossover cable thanks, ti ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
On Wed, Apr 04, 2007 at 09:55:33AM +0100, Chris Blunt wrote: Hi Tzafir / List. Thank you for your reply. I have run: make clean Configure Make Make install I get no compile errors, but still the same problems if I try to insmod zaptel As you suggested I tried modinfo zaptel Which resulted in: modinfo: could not find module zaptel Which suggests that the modules were installed to the wrong directory under /lib/modules . I also tried depmod with the same result and finally I tried insmod ./ztdummy from the src/zaptel-1.4.1 directory which resulted in: insmod: error inserting './ztdummy.ko': -1 Invalid module format This means that you built the modules vs. a kernel source tree that does not match your running kernel. What kernel do you run? What is the output of uname -a You mentioned you were running on CentOS. Do you have the proper kernel-devel package for your kernel? rpm -qa | grep kernel And while we're at it, let's check the first guess of the makefile for the location of the kernel source tree: ls -l /lib/modules/`uname -r` The build link there should have the information. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Job in Saudi Arabian Companies?
Hi, i need to find a job in Saudi Arabia related to the field of VoIP/Asterisk. But i live in Pakistan, so anyone who can provide me the list of companies working on VoIP/Asterisk related projects plz share. Thanx in advance -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P Need Help
Farooq Ahmed wrote: Hi All Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting follow message asterisk shows no port. Would be kind enough if somebody help me. Regards Farooq #misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Hi, in /etc/misdn-init.conf, switch the mode to te_ptmp= or something. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM: On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon now for the next DA question, where to find it (one)? Google has not been my friend. An alleged spot on sourceforge turned up blank. joe a. +++ www.j4computers.com 845-687-4563 Stone Ridge, NY 12484 +++ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] remote SIP, no audio, or one way audio.
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP - choppy sound on local LAN to T1
New install, Asterisk, obviously, Baystack 450 swtiches, verizon T1, Digium 4 port T1 card Some (few) users have had complaints from their clients that sound quality is poor. I do not know if the calls were placed via asterisk, or received via asterisk. If it matters. I believe this is a QoS issue, for the swtiches/infrastructure, wondering what can be done, if any one is familiar with these switches. If they are dog for Voip, recommendations? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
No I don't. So that will be my problem. Thanks. On 4/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Correct latency values in sip show peers
Rolz wrote: I was wondering if anyone knows how accurate the values are when you do a sip show peers from the CLI. My configuration is: Asterisk box (192.168.1.102) - gigabit switch - PC running x-lite (192.168.1.100) the CLI reports 101 ms delay however, ping is showing 1ms delay Where is the extra 100ms coming from? The softphone response? I'm not sure, but I think that ping isn't OSI ISO Layer 7, while softphone is. So, this 100 ms can come from there. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 15
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] make a call with IP address
Hello all, We are setting up a gateway in which the SIP devices will be connected dynamically using the Asterisk system. We use the originate Manager API command from our code to call an IP as (SIP/[EMAIL PROTECTED]). The call rings on the phone and goes through the normal (default) context and finally hangs up(WARNING[13833]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'GTW' ). Want we want to do it originate a simular call to another device say SIP/[EMAIL PROTECTED] and bridge the two connections. Can we expect some hints to move further to establish the call between SIP/1 and SIP/2. We are not interested in creating static entries in the .conf files, but open to use Manager API to build the system on-the-fly. All we want from the experts is that to validate our logic whether it is feasible to build up the communication system using the above technique and suggest us the best way to go about. regards, Pandi.P ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Wed, 4 Apr 2007, Joe Acquisto wrote: Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM: On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon now for the next DA question, where to find it (one)? Google has not been my friend. An alleged spot on sourceforge turned up blank. http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
. . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play blank sound while VM recording?
On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED] wrote: Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. Maybe this is what you need?: ;rtpkeepalive=secs; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) (in sip.conf, [general] section) Regards, Philipp That was exactly what I needed, thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Wed, 4 Apr 2007, Joe Acquisto wrote: . . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? No. All I should need is the linux daemon thing-let, correct? Yes. AIUI, You'll need 2 IP addresses to run it on though. Gordon joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
Joe Acquisto wrote: . . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The linux daemon is also downloadable there i think ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
On 4/4/07, Bruce Reeves [EMAIL PROTECTED] wrote:
You have a syntax error.
exten = _#78,n,Set(DB(${DND/CALLERID
(num)})=1)
should read
exten = _#78,n,Set(DB(DND/ ${CALLERID
(num)})=1)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hmmm...
Had hoped this would be easy, maybe still is, but running into a problem:
When I dial #78, I get a fast busy and these errors on the CLI
[Apr 4 00:39:29] ERROR[4046]: pbx.c:1523 ast_func_read: Function
DND/CALLERID not registered
[Apr 4 00:39:29] WARNING[4046]: func_db.c:87 function_db_write: DB
requires an argument, DB(family/key)=value
- - -
Here is the extensions.conf entry:
[dnd-on]
exten = _#78,1,Answer()
exten = _#78,n,Wait(1)
exten = _#78,n,Set(DB(${DND/CALLERID(num)})=1)
exten = _#78,n,Playback(do-not-disturb)
exten = _#78,n,Playback(enabled)
exten = _#78,n,Hangup()
- - -
It appears to me that Set(DB ... as a function isn't working, isn't
built in, or needs more information.
I saw something about GLOBAL variables, perhaps I can use those instead?
On 4/3/07, Doug Lytle [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
; **
; Do not disturb can be set via Asterisk
; instead of the phones by dialing this
; number.
; **
exten = 79*,1,Set(CALLBACK=${DB(DND/${CALLERIDNUM})})
exten = 79*,2,GotoIf($[${CALLBACK} = YES]?79*,3:79*,101)
exten = 79*,3,Set(DB(DND/${CALLERIDNUM})=NO)
exten = 79*,4,Playback(local/stutter)
exten = 79*,5,Playback(de-activated)
exten = 79*,6,Hangup()
exten = 79*,101,Set(DB(DND/${CALLERIDNUM})=YES)
exten = 79*,102,Playback(local/stutter)
exten = 79*,103,Playback(activated)
exten = 79*,104,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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--
Bruce Reeves
Nortex Networks
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[asterisk-users] Ring file
When I call into my Asterisk system, before the call is picked up, I hear a ring tone. What is that tone called where is stored and configured. I'd like to replace the ring with an announcement that is played until the call is picked up or put into voicemail. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: asterisk-users Digest, Vol 33, Issue 15
Hi Tzafir / List Here is some more information obtained from the commands you gave me: 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:21:39 CDT 2006 i686 i686 i386 GNU/Linux kernel-2.6.9-42.EL kernel-smp-2.6.9-42.EL kernel-ib-1.0-1 kernel-devel-2.6.9-42.0.3.EL kernel-2.6.9-42.0.3.EL kernel-smp-2.6.9-42.0.3.EL kernel-utils-2.4-13.1.83 I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Again thanks for your help, I am only a Linux beginner, and even more of a noob with CentOS. Best regards Chris -- Chris Blunt -Original Message- This means that you built the modules vs. a kernel source tree that does not match your running kernel. What kernel do you run? What is the output of uname -a You mentioned you were running on CentOS. Do you have the proper kernel-devel package for your kernel? rpm -qa | grep kernel And while we're at it, let's check the first guess of the makefile for the location of the kernel source tree: ls -l /lib/modules/`uname -r` The build link there should have the information. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 15
On Wed, Apr 04, 2007 at 02:31:54PM +0100, Chris Blunt wrote: Hi Tzafir / List Here is some more information obtained from the commands you gave me: 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:21:39 CDT 2006 i686 i686 i386 GNU/Linux kernel-2.6.9-42.EL kernel-smp-2.6.9-42.EL kernel-ib-1.0-1 kernel-devel-2.6.9-42.0.3.EL kernel-2.6.9-42.0.3.EL kernel-smp-2.6.9-42.0.3.EL kernel-utils-2.4-13.1.83 yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using DUNDi in a failover environment
Greetings list, There have been quite a few posts on the list over the last few months about using DUNDi to ensure users are always reachable even when logged into different asterisk boxes (as part of a load balancing cluster). For example, yesterday, this was in a post: (Olle Johansson) In combination with Dundi and the regexten= system, it's even more dynamic. Are there any documents/examples people have come across out there about using DUNDi to achieve load balancing/failover between 2 or more asterisk boxes? I've used DUNDi in the past, but primarily as a method of ensuring calls between locations take the lowest cost route (i.e. directly through the net rather than out as a PSTN call, across the PSTN, and back in at the other location). Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
It seems that it's already there (else you'd not get an error from
func_db)
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I do the same with:
exten = *49,n,Set(DB(${CALLERID(num)}/doNotDisturb)=1)
So I keep them the other way round.
Why not put in something like:
exten = _#78,n,Noop(DB(DND/${CALLERID(num)})=1)
and just see what it says? Maybe your callerId is somehow not being set
correctly?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
I'd definately check the caller-id - if it's a local phone on an FXS port,
then you might even want to hard-wire the caller-id in the
/etc/asterisk/zapata.conf file. eg.
; Channel 1: Local analogue line
context=internal
group=0
signalling=fxo_ks
sendcalleridafter=2
rxgain=3
txgain=3
mailbox=100
callerid=Shared DECT 100
channel = 1
Gordon
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[asterisk-users] disabling authentication
Is there a way to cause asterisk to accept all calls without any authentication? Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - Automatic Redial on No Answer
Hi, It seems to perfectly match what I was after : - Alice calls Bob and Bob doesn't answer (busy ? not there ?). - Alice hangs up and dials something (*41 for instance). - Whenever Bob is hanging up a call (that would prove Bob is back and probably available), a call from Alice to Bob is triggered. Is *41 called a vertical service activation code like here ? http://www.voip-info.org/wiki/view/Asterisk+vertical+service+activation+codes Anyway, I would be very pleased to look at your code. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring file
It can very well be that your TISP is playing that (if you are using voip). The way to test is to have Exten = s,1,Answer exten = s,2,Playback(tt-monkeys) If you still hear ringing before it plays the file then there isnt much that you can do. - Original Message - From: John Schmerold [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 04, 2007 4:18 PM Subject: [asterisk-users] Ring file When I call into my Asterisk system, before the call is picked up, I hear a ring tone. What is that tone called where is stored and configured. I'd like to replace the ring with an announcement that is played until the call is picked up or put into voicemail. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
Are you on a VPS ? - Original Message - From: Chris Blunt To: asterisk-users@lists.digium.com Sent: Wednesday, April 04, 2007 11:55 AM Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS Hi Tzafir / List. Thank you for your reply. I have run: make clean Configure Make Make install I get no compile errors, but still the same problems if I try to insmod zaptel As you suggested I tried modinfo zaptel Which resulted in: modinfo: could not find module zaptel I also tried depmod with the same result and finally I tried insmod ./ztdummy from the src/zaptel-1.4.1 directory which resulted in: insmod: error inserting './ztdummy.ko': -1 Invalid module format Your continued help is much appreciated. Chris Original Message Reads. Message: 8 Date: Tue, 3 Apr 2007 19:57:40 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Have you run 'make install'? What is the output of modinfo zaptel Any change if you run: depmod [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory insmod ./ztdummy.ko But it should fail (e.g: because zaptel is not loaded). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- Chris Blunt -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lithuania
Try the biz list. - Original Message - From: Mattias Andersson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 04, 2007 12:31 AM Subject: [asterisk-users] Lithuania Hi All! Maybe a little of topic. Bout coming from Sweden and needing to call Lithuania a lot am I wondering if anyone on the list could recommend a sheep service in Lithuania to connect my Asterisk to. A local number are not necessary bout preferd for incoming calls for my contacts. Regards Mattias Andersson Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql issue
Trying to create an extension that will toggle an enum value in our
database...
exten = s,1,MYSQL(Connect connid localhost myuser tmppass asterisk)
exten = s,n,MYSQL(Query resultid ${connid} UPDATE\ night_service\ SET\
status=(SELECT\ CASE\ status\ WHEN\ \'y\'\ THEN\ \'n\'\ ELSE\ \'y\'\
END));
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,Hangup
This is saying its exiting with a 0 value, and then I get a busy. Does
anyone see what I did wrong here? I'm sure its simple, just not to me :).
Rob
Why dont you use an agi ?
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[asterisk-users] Parked calls and Music on hold
Hello everybody, I'm trying to understand how can I set the MoH class for parked calls. I set the incoming class for calls, and it works correctly. When I park the call, the music on hold is ok, but when I close the communication on the parking side, the parked call gets the default music on hold class. Can someone explain me what's going on? Thanks in advance, -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Asterisk USER PORTAL
Do you know of any general User Portal applications for various Asterisk installations or are the Druid, Trixbox (sort of) etc all installation specific and not platform transferable? After looking yesterday this doesn't seem to exist. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation The current GUI built by Digium supports the basics of system administration, but this is only the beginning. The technology represents an unlimited extension to the current Asterisk model. Within a few months I suspect we will see a User Portal which offers full graphical control of features, a complete call history, realtime control of call recording, visual voicemail (do I have to pay Apple or Cingular for saying that?), on-screen call control, telephony-services-as-web-services and much more. From: Dean Collins Sent: Tuesday, 3 April 2007 9:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Asterisk USER PORTAL I'm trying to get the Mexuar development team to write some code to work with an existing asterisk USER PORTAL that presents a user with customized image of their Asterisk activities; * Address book, * Fop or some other kind of gui activity display * Voicemail access * Any other feature that should be integrated into a user portal page (maybe right click to dial or similar) We would like to develop some code that works with your existing User Portal to implement the Mexuar Corraleta IAX2 java applet softphone If you have a suitable portal or configuration or if you know of one I should be looking at can you please call me here in New York or email me some screen prints, if your portal is selected then this will give you additional functionality that you can market to your customers (we'll also throw in a license or two for you to set up as demo's for your clients). We basically just want to demonstrate this as a possible use for the Mexuar Corraleta technology on the demo pages of our website. URL links for you to check out; www.Mexuar.com http://www.mexuar.com/ www.Mexuar.com/Demo/Demo1 Flash Demo Page; http://www.mexuar.com/downloads/Level1Products/CorraletaDemoSound.swf Technical page for Asterisk System Integrators; http://www.voip-info.org/wiki/view/mexuar Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation www.Mexuar.com http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. image002.gif Description: image002.gif image003.gif Description: image003.gif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime
Hello, I'm using Realtime to select extensions out of a database so that we can provision inbound tollfree on the fly. Once I 'catch' the inbound, I want to get out of realtime and use the regular extensions again. I thought I could just use the goto statement and go to another context/entension in the 'non-realtime world'. Is this not possible? Is it all or nothing with Realtime? Thanks, Jason Wolfe It is possible. In fact you can have use a mix of real time and static and jump between the two even in the same context with real time. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 603 Error
- Original Message - From: Olle E Johansson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 03, 2007 8:59 AM Subject: Re: [asterisk-users] 603 Error 2 apr 2007 kl. 10.16 skrev Dovid B: Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP XXX.XXX.XX.XXX: 5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: sip:XXX.XXX.XX.XX;tag=a21dc3d8dd92817bo0 To: sip:XXX.XXX.XX.XX;tag=as7b187bff Call-ID: [EMAIL PROTECTED] CSeq: 112226 NOTIFY User-Agent: Blah Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Your server is sending a NOTIFY that the ITSP's server doesn't like. Propably a mailbox notification. Not a critical error, just a configuration issue. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! Thanks. Our lines were down so I was just guessing. Ended up being the ITSP's fault. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
I have created this before. I have to dig up the dial plan. The way I created it is it would call user1. User1 had the option to take the call, pass it to the next user or send it to VM. If he passed it to the next user, User2 had the same options as user1 and it flows down the list. Also every user has the option to opt in or out to receive calls or to have them just passed up (i.e. so that if user2 is busy then it should jump from user1 to user3). - Original Message - From: Andy Hester [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 02, 2007 8:15 PM Subject: RE: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT Andrew Joakimsen wrote: The logic of the macro is totally opposite of what it should be. I do recall sending a corrected version of the script to someone a while back, it might be on the mailing lists archive. However, there is an option for the Dial() command to do exactly what you wish p: This option enables screening mode. This is basically Privacy mode Thanks for the response - I missed that Dial option... Couple of questions on this: 1. I do not want to screen based on caller, instead I need to play the same message to a list of potential call recipients and allow each recipient to decide whether or not to accept the call based on whether or not they are available (for work for example). I understand that this option checks for a file. I will be transferring a call to this call coverage. How do I make sure that all the calls look for the same recording to play to the call screeners? 2. Does anyone have any dial plan examples of this type of set up? Thanks, -- Andy Hester Network Engineer Architel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Correct latency values in sip show peers
Tomislav Parcina wrote: Rolz wrote: I was wondering if anyone knows how accurate the values are when you do a sip show peers from the CLI. My configuration is: Asterisk box (192.168.1.102) - gigabit switch - PC running x-lite (192.168.1.100) the CLI reports 101 ms delay however, ping is showing 1ms delay Where is the extra 100ms coming from? The softphone response? I'm not sure, but I think that ping isn't OSI ISO Layer 7, while softphone is. So, this 100 ms can come from there. The times shown are the time to get a response to a SIP OPTIONS packet sent to the phone, not the time to get a response from an ICMP ECHO (ping) packet. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 16
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Channel: SIP/12345
Callerid: auto 12345
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notify
Extension: 1
Priority: 1
extensions.ael follows:
context notify {
1 = {
start:
Answer();
Wait(1);
Playback(ulii_01);
HangUp();
};
I want to dial number 12345 with taking into account the dial plan,
written in context.
when i'm trying to set:
Channel: SIP/[EMAIL PROTECTED]
asterisk say's:
chan_sip.c:2737 create_addr: No such host: context
attempt to set:
Channel: SIP/context/12345
has the same result
asterisk version is 1.4.2
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[asterisk-users] Which GUI for call screening ?
Hello, I'm wondering how it would be best for user to manage a whitelist-backlist of incoming calls to be screened : 1. Would you choose typical cases (allow-forbid internal-external calls) ? 2. Would you teach users regular expressions (so that then can receive calls from mobile) ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring sip.conf to allow guest access
Hi, I am configuring a conferencing server and need to allow SIP clients guest access. In iax.conf, I can allow guest access to the [conference] context with this entry === iax.conf == [guest] type=user host=dynamic context=conference So anyone connecting without username/password will be logged in as guest and restricted to the conference context. How can I do the same in sip.conf? How do I also configure my xlite client to login as guest? How to configure IAX is explained here http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf but not for SIP. http://www.voip-info.org/wiki-Asterisk+config+sip.conf Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring sip.conf to allow guest access
Tried this...it worked...but is this the best way? == sip.conf == [general] context=conference allowguest=yes [guest] type=friend nat=yes host=dynamic canreinvite=no context=conference --- Richard OSS [EMAIL PROTECTED] wrote: Hi, I am configuring a conferencing server and need to allow SIP clients guest access. In iax.conf, I can allow guest access to the [conference] context with this entry === iax.conf == [guest] type=user host=dynamic context=conference So anyone connecting without username/password will be logged in as guest and restricted to the conference context. How can I do the same in sip.conf? How do I also configure my xlite client to login as guest? How to configure IAX is explained here http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf but not for SIP. http://www.voip-info.org/wiki-Asterisk+config+sip.conf Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
Dovid B wrote: I have created this before. I have to dig up the dial plan. The way I created it is it would call user1. User1 had the option to take the call, pass it to the next user or send it to VM. If he passed it to the next user, User2 had the same options as user1 and it flows down the list. Also every user has the option to opt in or out to receive calls or to have them just passed up (i.e. so that if user2 is busy then it should jump from user1 to user3). Thanks, I'd be very interested to see this. -- Andy Hester Network Engineer Architel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemail to text translation)
(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed, 2007-04-04 at 09:25 -0700, [EMAIL PROTECTED] wrote: Date: Wed, 4 Apr 2007 09:25:02 -0700 From: Mike Taht [EMAIL PROTECTED] Subject: Re: [asterisk-dev] Voicemail to text translation To: [EMAIL PROTECTED],Asterisk Developers Mailing List asterisk-dev@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On 4/4/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is anybody aware of a way to automate the translation or conversion of voice mail files into text ? Being that understanding random human speech at 8khz I had had a different idea. Merely have a voice mail option press 4 to transcribe this - which would take the vmail and ship it to a transcription service like transcribr.com. There's a couple companies like that that out there do transcription - quite well, and cheaply. Sent via BlackBerry from T-Mobile ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Mike Taht PostCards From the Bleeding Edge http://the-edge.blogspot.com -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Console messages
Hi, how can I see in the console only my commands and its results? There is any way to disable the activity logger in the console by command? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Chis. Contact me off line if you are interested i can go to you system via ssh and then tell you what happend. Regards. Polo [EMAIL PROTECTED] -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Chris Blunt Enviado el: Miércoles, 04 de Abril de 2007 10:53 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Easiest method in a nutshell... iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] make Zaptel 1.2.16 'struct inode' has no member named 'u'.
On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: *** [_module_/usr/src/zaptel-1.2.16] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2257.fc5-smp-i686' make: *** [all] Error 2 An internet search has turned this message up but other than indicating that the inode structure has changed I'm no further ahead. I have found nothing specific for Asterisk. Any advice appreciated. Cameron ___ The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider. http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289
Just wanted to update the group. I copied the config file to a working server and the hints worked without any problems. Can anyone tell me if they have a working system using hits and SVN-branch-1.4-r59289 or newer. Eric Hall From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, April 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons 21 secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman 23 secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith 25 secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris 26 secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson 29 secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson 30 secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include = parkedcalls exten = 21,hint(SIP/21) exten = 21,1,answer exten = 21,n,dial(sip/21|30|kw) exten = 21,n,voicemail([EMAIL PROTECTED]|u) exten = 23,hint(sip/23) exten = 23,1,answer exten = 23,n,dial(sip/23|30|kw) exten = 23,n,voicemail([EMAIL PROTECTED]|u) exten = 25,hint(SIP/25) exten = 25,1,answer exten = 25,n,dial(sip/25|30|kw) exten = 25,n,voicemail([EMAIL PROTECTED]|u) exten = 26,hint(SIP/26) exten = 26,1,answer exten = 26,n,dial(sip/26|30|kw) exten = 26,n,voicemail([EMAIL PROTECTED]|u) exten = 29,hint(SIP/29) exten = 29,1,answer exten = 29,n,dial(sip/29|30|kw) exten = 29,n,voicemail([EMAIL PROTECTED]|u) exten = 30,hint(SIP/30) exten = 30,1,answer exten = 30,n,dial(sip/30|30|kw) exten = 30,n,voicemail([EMAIL PROTECTED]|u) -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.25/744 - Release Date: 4/3/2007 5:32 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
Oops, sorry. My explanation was not clear. I replaced that function in the codecs/Makefile in asterisk. I will attach the modified your patch. I hope this explanation helps. -- Xiu YuShen 2007/4/4, Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Apr 04, 2007 at 06:50:50PM +0900, Xiu YuShen wrote: 2007/3/17, Tzafrir Cohen [EMAIL PROTECTED]: http://bugs.digium.com/view.php?id=9303 Please test. I tested this patch to asterisk-1.2.17 with zaptel-1.2.10 and redhat 9. However, to compile on my environment, 'first' function was replaced by the 'firstword' function. Could you please be more specific? Replace where, exactly? In asterisk? In zaptel? In which source file? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ast_12_nozapnodec.diff.new Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call dies when I press *
Hi Mike - Well, when I restart the cli as requested below and go the addition steps of setting verbose to 25 and turning sip debug on for the phone in test, I don't get ANYTHING on the console. Sounds like it's a phone issue after all, right? I've got the same symptoms for BOTH the Sipura 2002 and a Polycom 501. Any ideas where to start? What is your dtmfmode set to in sip.conf (e.g. inband, rfc2833)? - Noah Thanx, Mike Diehl. On Thursday 29 March 2007 11:52, Doug wrote: At 18:23 3/28/2007, Mike Diehl wrote: Actually, it turns out that sometimes I can't get ANY DTMF to work. I can call a local phone number and log into my voicemail system at work. But my wife is unable to dial a toll free number and use their IVR. Hope this helps. What does your log read? asterisk -rvvv On Wednesday 28 March 2007 16:58, Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Diehl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] make Zaptel 1.2.16 'struct inode' has no member named 'u'.
Zaptel has no direct code relationship with Asterisk. Your error is because zaptel is trying to use a member no longer exists in newer kernels. However you are using fedora, and fedora included that change in older kernel. I found this in xpp/xbus-core.c /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use such a kernel, Change the * following test from 2,6,19 to 2,6,18. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif So go ahead and change the source as the comment says. Regards On 4/4/07, kjcsb [EMAIL PROTECTED] wrote: On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: *** [_module_/usr/src/zaptel-1.2.16] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2257.fc5-smp-i686' make: *** [all] Error 2 An internet search has turned this message up but other than indicating that the inode structure has changed I'm no further ahead. I have found nothing specific for Asterisk. Any advice appreciated. Cameron ___ The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider. http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
Ah!
Got it. Hard coding CallerID is a good idea and thank you for the example.
I decided to try the Noop(DB(...)) to see what was getting passed and
the empty CALLERID was the issue.
I decided to skip that and implement a global DND since that's what I
wanted anyway so I just set DND/ALL=1 in the DB line.
I'll post a full example here when I put on the finishing touches but
it is working now. Thanks all for the help.
One question... are there any places to get extra sound files like
activated or deactivated or do not disturb is... ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
On 4/4/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
It seems that it's already there (else you'd not get an error from
func_db)
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I do the same with:
exten = *49,n,Set(DB(${CALLERID(num)}/doNotDisturb)=1)
So I keep them the other way round.
Why not put in something like:
exten = _#78,n,Noop(DB(DND/${CALLERID(num)})=1)
and just see what it says? Maybe your callerId is somehow not being set
correctly?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
I'd definately check the caller-id - if it's a local phone on an FXS port,
then you might even want to hard-wire the caller-id in the
/etc/asterisk/zapata.conf file. eg.
; Channel 1: Local analogue line
context=internal
group=0
signalling=fxo_ks
sendcalleridafter=2
rxgain=3
txgain=3
mailbox=100
callerid=Shared DECT 100
channel = 1
Gordon
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 17
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speex codec in 1.4.2
Hello, I just upgraded my system from 1.2.10 to 1.4.2 Now I am having problems with speex codec. sound is totally garbled and destroyed. In 1.2.10 speex codec worked ok. As a SIP client I am using ekiga with narrowband speex (8000bps) enabled. Any ideas? Regards, Jure Petrovic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 18
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue application strategy
I am using rrmemory for my queues. I have noticed that the application will only distribute or dial one number at a time. Is there a different strategy that will allow the queue to distribute more than one call at a time? I don't want to use ringall because that would tie up thirteen of my trunks every time it tried to distribute a call. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tunnel Q.SIG through an IP network
Hi, Today's setup is : Legacy PBX1 with E1 --- Leased line - Legacy PBX2 with E1 Prospective setup is : PBX1 --- Asterisk GateWay1 (with Digium E1) -- IP network -- Asterisk GW2 (with Digium E1) - PBX2 Is there a way to tunnel, transport or translate Q.SIG signals between both PBXs ? The IP network is just used for point to point leased lines replacement : failover or other fancy features are not required. As I'm not sure Asterisk has features to fully understand Q.SIG spoken between legacy PBX, I thought simple transportation (HDLC ? SS7 then IP ?) would be enough. Any idea or comment ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
Easiest method in a nutshell... iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Sorry, this is intended to do what for me? I cannot find -j in the iptables man page. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call dies when I press *
There wasn't a setting, but I set it to rfc2833. On Wednesday 04 April 2007 12:49, Noah Miller wrote: Hi Mike - Well, when I restart the cli as requested below and go the addition steps of setting verbose to 25 and turning sip debug on for the phone in test, I don't get ANYTHING on the console. Sounds like it's a phone issue after all, right? I've got the same symptoms for BOTH the Sipura 2002 and a Polycom 501. Any ideas where to start? What is your dtmfmode set to in sip.conf (e.g. inband, rfc2833)? - Noah Thanx, Mike Diehl. On Thursday 29 March 2007 11:52, Doug wrote: At 18:23 3/28/2007, Mike Diehl wrote: Actually, it turns out that sometimes I can't get ANY DTMF to work. I can call a local phone number and log into my voicemail system at work. But my wife is unable to dial a toll free number and use their IVR. Hope this helps. What does your log read? asterisk -rvvv On Wednesday 28 March 2007 16:58, Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Diehl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Diehl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
Joe Acquisto [EMAIL PROTECTED] Wrote: 4/4/2007 4:24 PM: Easiest method in a nutshell... iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Sorry, this is intended to do what for me? I cannot find -j in the iptables man page. joe a. I found the -j but am still unclear what this does for the dropped audio issue? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
One question... are there any places to get extra sound files like
activated or deactivated or do not disturb is... ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
As I couldn't find such files under 'sounds', I created them by hand. It's
easy, only if you have a few minutes:
1. Download and install a free mp3/wma recorder - I found
http://www.xaudiotools.com/ to be quite handy
2. Record the required sounds - I chose .wav format with PCM 8000khz; 16bit;
Mono for two reasons: First, the file size was small so i could ftp fast
back to the server. Second, the sound quality after converting to .gsm was
fantastic.
3. convert .wav to .gsm using sox:
$ sox soundfile.wav soundfile.gsm
4. copy the file back to sounds directory
- Original Message -
From: Brian McEntire [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 04, 2007 2:52 PM
Subject: Re: [asterisk-users] Adding DND to dialplan
Ah!
Got it. Hard coding CallerID is a good idea and thank you for the example.
I decided to try the Noop(DB(...)) to see what was getting passed and
the empty CALLERID was the issue.
I decided to skip that and implement a global DND since that's what I
wanted anyway so I just set DND/ALL=1 in the DB line.
I'll post a full example here when I put on the finishing touches but
it is working now. Thanks all for the help.
One question... are there any places to get extra sound files like
activated or deactivated or do not disturb is... ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
On 4/4/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
It seems that it's already there (else you'd not get an error from
func_db)
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I do the same with:
exten = *49,n,Set(DB(${CALLERID(num)}/doNotDisturb)=1)
So I keep them the other way round.
Why not put in something like:
exten = _#78,n,Noop(DB(DND/${CALLERID(num)})=1)
and just see what it says? Maybe your callerId is somehow not being set
correctly?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
I'd definately check the caller-id - if it's a local phone on an FXS
port,
then you might even want to hard-wire the caller-id in the
/etc/asterisk/zapata.conf file. eg.
; Channel 1: Local analogue line
context=internal
group=0
signalling=fxo_ks
sendcalleridafter=2
rxgain=3
txgain=3
mailbox=100
callerid=Shared DECT 100
channel = 1
Gordon
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[asterisk-users] Polycom
I know this doesn't belong on this list but... I am looking to see if anyone is using Polycom and knows of a web based software for creating/managing the cfg files for polycom phones. I see that the AsteriskNow will add provisioning support for Polycom phones. Since it is still in beta, I was just looking to see if there was anything else out there. Thanks! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue application strategy
If you are using Asterisk 1.4 you should look at the autofill configuration option in queues.conf. For versions prior to that, I'm not sure there is a solution. On 4/4/07, Jordan Novak [EMAIL PROTECTED] wrote: I am using rrmemory for my queues. I have noticed that the application will only distribute or dial one number at a time. Is there a different strategy that will allow the queue to distribute more than one call at a time? I don't want to use ringall because that would tie up thirteen of my trunks every time it tried to distribute a call. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pound # key not being handled
I am trying to use call parking. I have the following in features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? I am using asterisk-1.2.17 Thanks, Alberto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad Line Noise over T1
I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9 where is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible line noise in the earpiece of the softphone and low/distored audio back out of it to the hard phone on the Opt 11. The noise starts the moment the softphone goes 'off-hook' and the hard extension starts ringing and only gets worse once the call is picked up. (essentially white noise by then) Initially, during ringing, the noise is pulsed... like when a cell phone is next to speakers and there is no conversation on the line. Once the call is connected, the noise essentially fills the gaps in the audio... almost like comfort noise gone psycho. The noise reminds me of a modem when the carrier has been established between two endpoints... that static sound. Except that it clearly stops when there is audio on the channel. I've checked the error counters on the A101 card before and after a call and they look fine so it doesn't seem to be jitter or slip or anything like that on the T1. I also tried the calls with 'echocancel' 'echocancelwhenbridged' set to both yes no in Zapata.conf. I've also turned echo cancellation on and off on the A101 card using wancfg. I've tried txgain and rxgain values from 0.0 to -10.0 with no affect. Now, to really mess with things, if I dial another SIP softphone extension on the Asterisk box (or IVR or VM), the audio is pristine so I can rule out softphone problems and issues with the audio hardware on the PC. (Plus, I've tested this from several softphones and they all exhibit the problem.) It is only when I'm routing a call over that T1 that I get the noise. And to add to the mystery, the hard extension on the Opt 11 has no noise on the line. If I talk into the hard phone, I can hear it on the softphone perfectly but the noise fills all the gaps in the audio. If I talk into the softphone, I can hear it on the hard phone but the audio is a bit soft and distorted. I'm stumped on this. I've never ran into this type of audio problem before. Has anyone seen this before and found a solution? Below is Zapata.conf and Zaptel.conf Thanks! Jason ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en group = 0 context=from-zaptel signalling=pri_net switchtype = 5ess callerid = asreceived channel = 1-23 rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;group=1 ;Include AMP configs #include zapata_additional.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us # PRI to Nortel span=1,0,0,esf,b8zs bchan=1-23 dchan=24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
On Wed, 4 Apr 2007, Brian McEntire wrote: One question... are there any places to get extra sound files like activated or deactivated or do not disturb is... ?? I didn't find them in the sounds directory after a vanilla install of the latest stable asterisk 1.4. They are in the asterisk-sounds package. gordon @ yakko: ls *act* activated.gsm de-activated.gsm etc. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding DND to dialplan
Brian McEntire wrote: One question... are there any places to get extra sound files like activated or deactivated or do not disturb is... ?? I didn't find them in the sounds directory after a vanilla install of the latest stable asterisk 1.4. Maybe the asterisk-sounds tarball has some you could use. http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pound # key not being handled
Alberto Alonso wrote: I am trying to use call parking. I have the following in features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? See the examples in features.conf. Adjust the entries in [featuremap] and [applicationmap] and do something like Set(DYNAMIC_FEATURES=parkcall) in extensions.conf. Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
On Wed, 04 Apr 2007, Joe Acquisto wrote: iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Dur... that should have been -j ACCEPT. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and Asterisk
Well I would wonder how Polycom even had any idea whom your vendor is. On 4/2/07, Stephen Bosch [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: First-sale doctrine, unless your vendor did something illicit to obtain Polycom phones there is nothing they can do about it. What they can do is refuse to keep supplying the vendor, and that's a threat the vendors tend to take seriously, especially if the product is any good. This vendor certainly did, and warned me not to do it again. It's good to know Polycom has anti-competitive business practices. I also dislike that they refuse to give out anything but old firmware versions too. They could do a lot to improve their relationships with their public :( -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Asterisk-Addon-1.4.0 MySQL module
Hi Phillip, Thanks for replying. I do have all the item you listed in below email perviously. I reformatted my machine with FC5 this time and loaded up Asterisk 1.4.2with Asterisk-Addon 1.4 with MySQL modules now. I am sure the problem is related to FC6. I was pulling my hair out for a while lol.. Sincerely, K On 4/3/07, Philipp Kempgen [EMAIL PROTECTED] wrote: KC wrote: I still can't figure out why res_config_mysql module not showing up with many attempt. Anyone have any idea on this? checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts Sincerely, K -Original Message- From: KC [mailto:[EMAIL PROTECTED] Sent: Friday, March 30, 2007 1:43 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk-Addon-1.4.0 MySQL I can't find anything about Asteirsk-Addon-1.4 MYSQL problem from googling. I thought it would be my error but surely not just tried asterisk 1.2.17 with addon 1.2.5 and it work. Does anyone else having problem to make res_config_mysql, cdr_addon_mysql and app_addon_sql_mysql in addon-1.4? Thanks for sharing There are no res_config_mysql and cdr_addon_mysql module after. /configure make all make install in asterisk module directory. It would be great if someone can give me some hint. I never experienced this before with 1.2 releases. Is there something changed on 1.4 releases? Or am I missing something. I am about to pull my hair out after many hours looking at the monitor. uname -a Linux 2.6.20-1.2933.fc6 #1 SMP Mon Mar 19 10:42:48 EDT 2007 i686 i686 i386 GNU/Linux rpm -qa | grep -i mysql mysql-5.0.27-1.fc6 php-mysql-5.1.6-3.4.fc6 mysql-devel-5.0.27-1.fc6 perl-DBD-MySQL-3.0007-1.fc6 mysql-server-5.0.27-1.fc6 *CLI core show version Asterisk 1.4.2 on a i686 running Linux on 2007-03-28 05:45:27 UTC Did you install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf libtool make automake automake14 automake15 automake16 automake17 bison byacc flex libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel mysqlclient10 mysqlclient10-devel mysqlclient12 mysqlclient12-devel mysqlclient14 mysqlclient14-devel ? Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
After recompiling zaptel, did you recompile Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram kortleven Sent: Wednesday, April 04, 2007 14:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] misdn and debian
2007/4/3, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Apr 02, 2007 at 08:30:57PM +0300, Giedrius Augys wrote: Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near Apache2 starting I started my system with recovery kernel, and tun off misd, then my system works fine. I think it's problem with memory. Have you tried memtest? apt-get install memtest86 , enable it in /etc/grub/menu.lst and run 'update-grub' . Has anybody debian and misdn working fine? Maybe you can advices , what kernel and misdn versions to use... Let's think: what comes shortly after apache? maybe asterisk? to get a better idea: ls /etc/rc2.d This ialso suggests that you use asterisk from your own build rather than from the package. In the package asterisk starts after most other services, in order for the service asterisk to start after the service zaptel. Is asterisk running with the option '-p'? If so: disable it for the purpose of testing. It makes an asterisk 100% CPU loop into a hanged system. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have installed 2.6.20.1 kernel from source and misdn drivers (i have marked 'Support modular ISDN driver', 'Support for AVM FRITZ!Cards', digital audio, loop device,isdn tunnel ) on Debian testing machine. The new kernel compiles successfully. But when I enter command 'misdn-init start', my computer hangs up, and I need reset it... Please sugest me what to do, cause I work about 3 nights and stuck with misdn... This my message log: Apr 5 01:10:09 asterisk kernel: mISDN: DSS1 Rev. 1.47 Apr 5 01:10:09 asterisk kernel: mISDN Capi 2.0 driver file version 1.21 Apr 5 01:10:09 asterisk kernel: AVM Fritz PCI/PnP driver Rev. 1.43 Apr 5 01:10:09 asterisk kernel: kobject_add failed for fcpci with -EEXIST, don't try to register things with the same name i Apr 5 01:10:09 asterisk kernel: [c0201fcd] kobject_add+0x160/0x189 Apr 5 01:10:09 asterisk kernel: [c02020b6] kobject_register+0x19/0x30 Apr 5 01:10:09 asterisk kernel: [c026056d] bus_add_driver+0x4d/0x15f Apr 5 01:10:09 asterisk kernel: [c020d150] __pci_register_driver+0x64/0x90 Apr 5 01:10:09 asterisk kernel: [d88da0aa] Fritz_init+0xaa/0xe1 [avmfritz] Apr 5 01:10:09 asterisk kernel: [c013acdf] sys_init_module+0x1770/0x18b7 Apr 5 01:10:09 asterisk kernel: [c0102d58] syscall_call+0x7/0xb Apr 5 01:10:09 asterisk kernel: === Apr 5 01:10:09 asterisk kernel: mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) Apr 5 01:10:09 asterisk kernel: mISDN_dsp: DSP clocks every 128 samples. This equals 16 jiffies. Apr 5 01:10:09 asterisk kernel: d8a34b77 Apr 5 01:10:09 asterisk kernel: Modules linked in: mISDN_dsp mISDN_capi l3udss1 mISDN_l2 mISDN_l1 capi capifs kernelcapi mIS Apr 5 01:10:09 asterisk kernel: 6Adding 779144k swap on /dev/hda3. Priority:-1 extents:1 across:779144k Apr 5 01:10:09 asterisk kernel: EXT3 FS on hda1, internal journal Apr 5 01:10:09 asterisk kernel: loop: loaded (max 8 devices) Apr 5 01:10:09 asterisk kernel: device-mapper: ioctl: 4.11.0-ioctl(2006-10-12) initialised: [EMAIL PROTECTED] Apr 5 01:10:09 asterisk kernel: e100: eth0: e100_watchdog: link up, 100Mbps, full-duplex Apr 5 01:10:10 asterisk kernel: input: Power Button (FF) as /class/input/input2 Apr 5 01:10:10 asterisk kernel: ACPI: Power Button (FF) [PWRF] Apr 5 01:10:10 asterisk kernel: input: Power Button (CM) as /class/input/input3 Apr 5 01:10:10 asterisk kernel: ACPI: Power Button (CM) [PBTN] Apr 5 01:10:10 asterisk kernel: NET: Registered protocol family 10 Apr 5 01:10:10 asterisk kernel: lo: Disabled Privacy Extensions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
On Wed, Apr 04, 2007 at 11:51:21PM +0200, bram kortleven wrote: Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve Do you use the packages from Experimental? If so: indeed some of them were lacking chan_zap.so . Hwever that shoul have been fixed on the latest version (through a build-dependency on zaptel = 1.4.1 ) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 19
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no gtalk capable clients to talk to.
I am getting the message chan_gtalk.c:853 gtalk_alloc: no gtalk capable clients to talk to. What does it mean? How can I find or make a gtalk capable client? It is Asterisk SVN-trunk-r59043. Jabber show connected shows 1 connected jabber user. Jabber debug periodically shows JABBER: my_gtalk_account INCOMING: ___ Do You Yahoo!? La mejor conexión a Internet y b 2GB/b extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
Yes I did, I even started all over again using the original source tar.gz packages... The menuselect thing bothers me the most... something is wrong or at least, missing. When compiling, I see all chan_... things passing, but no chan_zap... Hate it ... Thanks for the help - Original Message: After recompiling zaptel, did you recompile Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram kortleven Sent: Wednesday, April 04, 2007 14:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070404/de4d4c46/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
J. Oquendo [EMAIL PROTECTED] Wrote: 4/4/2007 5:58 PM: On Wed, 04 Apr 2007, Joe Acquisto wrote: iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Dur... that should have been -j ACCEPT. Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Playback Issue
Ever since upgrading from 1.2.X to 1.4.2, I'm having trouble with voicemail. When played back, the messages start out okay, but after 10 seconds or so, the playback speed starts to increase and the voice becomes illegible. It seems like some kind of audio timing problem. Phone calls seem okay, in general. I don't have any digium cards, but I am using ztdummy. Ztdummy is loaded properly, and zttest gives be 99+% ratings. zap show status in the asterisk console shows that ZTDUMMY is loaded into asterisk. I've gone into the /var/spool/asterisk directory and played the WAV files directly, and I hear the same problem. So, I'm guessing the problem is introduced when the voicemail is recorded. Can anyone offer a suggestion as to what I can debug? Thanks, Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and Asterisk
Serial number? Andrew Joakimsen wrote: Well I would wonder how Polycom even had any idea whom your vendor is. On 4/2/07, Stephen Bosch [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: First-sale doctrine, unless your vendor did something illicit to obtain Polycom phones there is nothing they can do about it. What they can do is refuse to keep supplying the vendor, and that's a threat the vendors tend to take seriously, especially if the product is any good. This vendor certainly did, and warned me not to do it again. It's good to know Polycom has anti-competitive business practices. I also dislike that they refuse to give out anything but old firmware versions too. They could do a lot to improve their relationships with their public :( -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Using DUNDi in a failover environment
Are there any documents/examples people have come across out there about using DUNDi to achieve load balancing/failover between 2 or more asterisk boxes? I've used DUNDi in the past, but primarily as a method of ensuring calls between locations take the lowest cost route (i.e. directly through the net rather than out as a PSTN call, across the PSTN, and back in at the other location). This may help, using Dundi and SIP Realtime. http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_ Whitepaper.pdf I've been using this setup for months, if a registration server fails, I can still call the SIP phone by dipping the database for the full contact info. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
