Re: [asterisk-users] Call terminates after 20 minutes
Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: Jim Boykin pisze: We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Sesion-timer set but not supported by sip-peers? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank loop current adjustment
On Sat, Oct 25, 2008 at 11:53:21AM +0200, Tzafrir Cohen wrote: On Thu, Oct 23, 2008 at 02:50:05PM +0200, Udo Schacht-Wiegand wrote: For a door opener on an Astribank FXS port we need a loop current of 24.5mA . It does not function with the Astribank now, the dialtone becomes quiet immediately after pressing the button on that device. I've seen a limit of 23mA in the zaptel source. Is it possible to change the loop current of the Astribank somehow? A phone that is on-hook is supposed to be an open circuit and ideally not draw any current. In the real world this doesn't really happen. The Astribank detects this by setting two limits. Anything above the high limit is considered closed circuit (off-hook). Anything below the lower limit is considered open circuit (on-hook). This (and what followed) was a good answer to the wrong question (as he asked about the current when the line os off-hook). Luckily Udo insisted and was later answered in private mail. Anyway, such questions should normally go to [EMAIL PROTECTED] . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Media gateway controller
Hello everyone, I am new in voip. I want to use a linux pc as a media gateway controller (with Megaco protrocol if possible). I heard Asterisk could do it, but in the documentation I haven't found information about it. Could someone help me? Thank you very much. Pedro Gonzalez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regarding keywords in sip.conf/users.conf
Hi guys, I'm about to embark on a small (undoubtedly to get much larger) project to write a set of scripts to handle provisioning of phones - Snom to begin with, possibly with others (most likely Polycom and Linksys) to follow later. Since I want this script to handle *all* aspects of phone provisioning (such as BLF buttons and so on) I need a place to store data. My preference is to keep all phone related configuration in the one place - such as sip.conf or users.conf. How would having additional keywords (most likely with a prefix of some type to reduce the likelihood of conflicts with real keywords) in Asterisk's .conf files affect Asterisk? I would expect that Asterisk should ignore unknown keywords, but I'd rather check on this with those in the know first. Any insights? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ztdummy and Asterisk
Ideas are welcome. Thanks in advance. Aldo Hi there, Try running genzaptelconf -v and restart asterisk. Here's what I have in /etc/zaptel.conf in order to have the MeetMe working. # Autogenerated by /usr/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 (source: RTC) 1 (MASTER) # Global data loadzone= us defaultzone = us --- HTH Dobry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ztdummy and Asterisk
On Sun, Nov 02, 2008 at 05:30:15PM +0200, Dobry Dobrev wrote: Ideas are welcome. Thanks in advance. Aldo Hi there, Try running genzaptelconf -v and restart asterisk. Here's what I have in /etc/zaptel.conf in order to have the MeetMe working. -v doesn't change the output . zaptel.conf is not needed for ztdummy, as ztdummy does not handle spanconfig, start or whatever. It simply starts operating at the moment it is probed. # Autogenerated by /usr/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 (source: RTC) 1 (MASTER) # Global data loadzone= us defaultzone = us This thing might have been useful for zaptel channels, if you had any. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( On Sat, Nov 1, 2008 at 1:22 PM, Emmanuel Pascal Bruno [EMAIL PROTECTED]wrote: Oh ok, I knew it was something like that. I have tried many different settings on my router. I'll dig into it some more. Thanks On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis [EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this You likely have firewall issues since it appears that you are not receiving a response from the other end. Make sure you have *both* your SIP and RTP ports forwarded to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
Emmanuel Pascal Bruno wrote: I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( Disabling firewalls is almost certainly going to ensure the problem persists. You need to ensure that all SIP and RTP ports are port-forwarded from your firewall to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call terminates after 20 minutes
Any help. Thanks On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin [EMAIL PROTECTED] wrote: Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: Jim Boykin pisze: We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Sesion-timer set but not supported by sip-peers? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ztdummy and Asterisk
Hi to all, Thank you Dobry an Tzafrir for your answers. I agree with Tzafrir in the sense that ztdummy needs no configuration. Moreover, the .conf file generated by genzaptelconf does not contain any configuration at all for ztdummy (tonezone and defaultzone are the only uncommented text in it). I must also say that I have tried again with the current versions of Asterisk and Zaptel, with the same result. So I'm still wondering where the problem lies... Cordially, Aldo Hi there, Try running genzaptelconf -v and restart asterisk. Here's what I have in /etc/zaptel.conf in order to have the MeetMe working. -v doesn't change the output . zaptel.conf is not needed for ztdummy, as ztdummy does not handle spanconfig, start or whatever. It simply starts operating at the moment it is probed. # Autogenerated by /usr/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 (source: RTC) 1 (MASTER) # Global data loadzone= us defaultzone = us This thing might have been useful for zaptel channels, if you had any. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. What device? I am using TE412P. No message on the console of the machine? Yes, nothing at all. The machine just froze and had to be rebooted. This probably means one of two things: 1. Bad kernel-level deadlock (maybe caused by Zaptel) I will upgrade zaptel to the latest version. 2. If asterisk is running with -p: it might be in a 100% CPU loop. I just use whatever it is in /etc/init.d/asterisk. I checked the file and it does not come with a -p option. I checked /var/log/asterisk but there is nothing unusual I can see. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP traffic shaping
On 11/1/08, OCG Technical Support [EMAIL PROTECTED] wrote: This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- It was based off Wondershaper originally, enhanced for VoIP traffic and gives the option to use HFSC or HTB. Not only do I use it myself for AstLinux and Star2Star, most of the reports I've (we've) had have been favorable. Try it out! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf
It should ignore the keywords, but you will get lots of errors in the CLI. My guess is that if you put it all in a DB (and use realtime) you can probably do whatever you want. PaulH Rob Hillis wrote: Hi guys, I'm about to embark on a small (undoubtedly to get much larger) project to write a set of scripts to handle provisioning of phones - Snom to begin with, possibly with others (most likely Polycom and Linksys) to follow later. Since I want this script to handle *all* aspects of phone provisioning (such as BLF buttons and so on) I need a place to store data. My preference is to keep all phone related configuration in the one place - such as sip.conf or users.conf. How would having additional keywords (most likely with a prefix of some type to reduce the likelihood of conflicts with real keywords) in Asterisk's .conf files affect Asterisk? I would expect that Asterisk should ignore unknown keywords, but I'd rather check on this with those in the know first. Any insights? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call terminates after 20 minutes
Copy your dialplan and sip debug a call. On Sun, Nov 2, 2008 at 3:07 PM, Jim Boykin [EMAIL PROTECTED] wrote: Any help. Thanks On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin [EMAIL PROTECTED] wrote: Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: Jim Boykin pisze: We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Sesion-timer set but not supported by sip-peers? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan E. RodrÃguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 430 no hangup after SIP BYE, Status 481 instead
Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the call has ended. Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the phone but the phone responds with: Status 481 Call Leg/Transaction does not exist. The Polycom is currently associated with 2 sip servers (using 2 lines on the phone) because I am currently in the progress of migrating from one server to another. So the asterisk server is having issues with is on Line 2 and it works perfectly well on Line 1 with a completely different Asterisk server running 1.4.16.2. I haven't tried switching the lines around to see if its just a problem with it being on Line 2. The Polycom is running the latest Bootrom and Sip version. Does anyone have any idea what could be causing this? Cheers, -Joel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call problems
I have tried that too with no results On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis [EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( Disabling firewalls is almost certainly going to ensure the problem persists. You need to ensure that all SIP and RTP ports are port-forwarded from your firewall to your Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and bigmem kernel
Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I had to rmmod zaptel to get asterisk running. (2) SIP doesn't work Has anybody worked successfully with this kernel?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and bigmem kernel
On Sun, Nov 02, 2008 at 10:03:12PM -0600, Edgar Guadamuz wrote: Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I had to rmmod zaptel to get asterisk running. lsmod | grep ^zaptel zttest -c 3 (2) SIP doesn't work What did you do? What did you expect to happen? What actually happened? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
