Re: [asterisk-users] Dahdi 2.4.0 and unplugged spans [SOLVED]
2010/12/1 Shaun Ruffell sruff...@digium.com On 12/01/2010 11:00 AM, Shaun Ruffell wrote: On 12/01/2010 09:56 AM, Olivier wrote: I'm facing an issue with which loading wctdm24xxp module fails. [ 33.527942] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 33.527995] wctdm24xxp :01:0b.0: The firmware may be corrupted. Please completely power off your system, power on, and then reload th e driver with the 'forceload' module parameter set to 1 to attempt recovery. [ 33.544741] ACPI: PCI interrupt for device :01:0b.0 disabled [ 33.544757] wctdm24xxp: probe of :01:0b.0 failed with error -5 The essential thing here is that your Hx8 board doesn't appear to be responding. I've seen certain systems where revision 9397 [1] works around cases like what you're seeing. If that doesn't help your best bet is to contact Digium technical support for help with troubleshooting. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 I forgot to add, please update to the current trunk and retry as opposed to checking out revision 9397 specifically. ]# svn co http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux-trunk ]# cd dahdi-linux-trunk ]# make install -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I tried using revision 9507 (which currently matches trunk). At first, I still got the same errors in dmesg but, after a reboot, it seems to pass ! (The dark side of things is I still don't understand how to write a dahdi installation script which could run without rebooting but that's another story). Anyway this thread topic was to check if I needed a live PSTN connection to configure this HA8 board and the answer is no. Thanks for helping -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Any comments on Comcast and Level 3 story this week?
Hi, Lots of news about Comcast and Level 3 this week, here's a sample: http://vuc.li/hUsQrd We're hoping to attract some knowledgeable participants - that's why I'm posting here - to chat about this on the VUC. Please consider joining us to contribute what you know, starting sometime between 12:30 and 1PM EST Friday. For local time: http://vuc.me/next To connect with the VUC, one address for all the info: http://vuc.me Hope to see youthetre. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rotate of logfiles
Hello list. This is not a life-threatening question, but still quite important for debugging. I have the following crontab : 15 0,8,12,17 * * * /usr/sbin/asterisk -rx 'logger rotate' Because I have debug level 9, logfiles get quite large. I notice that the rotation of the logfiles goes to plan, except at 17h15. I currently have : -rw-r--r-- 1 root root 59024 Dec 2 09:36 messages.vps.hosting -- 8h15 till now -rw-r--r-- 1 root root 0 Dec 2 02:02 messages.vps.hosting.0 -- empty -rw-r--r-- 1 root root 54137 Dec 2 08:14 messages.vps.hosting.1 -- 0h15 till 8h15 -rw-r--r-- 1 root root283286 Dec 1 17:14 messages.vps.hosting.2 -- 12h15 till 17h15 -rw-r--r-- 1 root root255762 Dec 1 12:14 messages.vps.hosting.3 -- 8h15 till 12h15 -rw-r--r-- 1 root root 0 Dec 1 02:02 messages.vps.hosting.4 -- empty -rw-r--r-- 1 root root 6 Dec 1 08:12 messages.vps.hosting.5 -- 0h15 till 8h15 -rw-r--r-- 1 root root315654 Nov 30 17:14 messages.vps.hosting.6 -- 12h15 till 17h15 -rw-r--r-- 1 root root290819 Nov 30 12:13 messages.vps.hosting.7 -- 8h15 till 13h15 -rw-r--r-- 1 root root 0 Nov 30 02:02 messages.vps.hosting.8-- empty -rw-r--r-- 1 root root 44291 Nov 30 08:08 messages.vps.hosting.9 -- 0h15 till 8h15 I'm missing a logfile covering 17h15 till 0h15... and I have empty logfiles... Using asterisk 1.6.2.10. Thanks for your feedback. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rotate of logfiles
On Thu, Dec 02, 2010 at 09:44:50AM +0100, Jonas Kellens wrote:
Hello list.
This is not a life-threatening question, but still quite important for
debugging.
I have the following crontab :
15 0,8,12,17 * * * /usr/sbin/asterisk -rx 'logger rotate'
Because I have debug level 9, logfiles get quite large.
I notice that the rotation of the logfiles goes to plan, except at 17h15.
I currently have :
-rw-r--r-- 1 root root 59024 Dec 2 09:36 messages.vps.hosting
-- 8h15 till now
-rw-r--r-- 1 root root 0 Dec 2 02:02 messages.vps.hosting.0
-- empty
-rw-r--r-- 1 root root 54137 Dec 2 08:14 messages.vps.hosting.1
-- 0h15 till 8h15
-rw-r--r-- 1 root root283286 Dec 1 17:14 messages.vps.hosting.2
-- 12h15 till 17h15
-rw-r--r-- 1 root root255762 Dec 1 12:14 messages.vps.hosting.3
-- 8h15 till 12h15
-rw-r--r-- 1 root root 0 Dec 1 02:02 messages.vps.hosting.4
-- empty
-rw-r--r-- 1 root root 6 Dec 1 08:12 messages.vps.hosting.5
-- 0h15 till 8h15
-rw-r--r-- 1 root root315654 Nov 30 17:14 messages.vps.hosting.6
-- 12h15 till 17h15
-rw-r--r-- 1 root root290819 Nov 30 12:13 messages.vps.hosting.7
-- 8h15 till 13h15
-rw-r--r-- 1 root root 0 Nov 30 02:02 messages.vps.hosting.8
-- empty
-rw-r--r-- 1 root root 44291 Nov 30 08:08 messages.vps.hosting.9
-- 0h15 till 8h15
I'm missing a logfile covering 17h15 till 0h15... and I have empty
logfiles...
Using asterisk 1.6.2.10.
Why not use the standard logrotate package? My
/etc/logrotate.d/asterisk:
/var/log/asterisk/debug /var/log/asterisk/messages
/var/log/asterisk/full /var/log/asterisk/*_log {
weekly
missingok
rotate 4
sharedscripts
postrotate
/usr/sbin/invoke-rc.d asterisk logger-reload /dev/null
2 /dev/null
endscript
}
Note the postrotate scriptlet.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Zaptel / Asterisk on Solaris
On Wednesday 01 Dec 2010, RR wrote: Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met. This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which according to some rigorous mathematical calculations, I concluded IS better than v5.7. Assuming Solaris is anything like Linux, the installer will just be a shell script. Open the script in a text editor and search for the text of the error message. It will be wrapped inside an `if` statement, just alter this so the test always passes. I had to do something similar to allow the Flashplayer installer to install the 32-bit Flash binary into users' home directories held on a 64-bit NFS server and exported to 32-bit workstations, right from the server. -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Thu, Dec 2, 2010 at 4:15 AM, bilal ghayyad bilmar...@yahoo.com wrote: Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi on Realtime.
There is no specific Realtime database for chan_dahdi (that I know if). You can store the configuration using Realtime Static using the new chan_dahdi.conf notation without any problems. The only problem with Realtime Static is that you cannot use the text file, you need to load everything from the database. Another possibility would be to use an #exec from chan_dahdi.conf to extract the channel configuration from the database. Thanks for the reply Carlos. You have the model of the tables for chan_dahdi in static mode? This quite difficult to find on the internet ... And you know if the generals can also be included in a static way? Thanks again At, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Push central phone book to phones
Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Any idea what is it about SIP over IAX2 that made such an improvement? -M On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, December 02, 2010 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Push central phone book to phones Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. First impression is that it is doubtful. Your best bet to do this would be an xml file if all 4 types will read that as a phone book/directory. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, 2 Dec 2010, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Grandstreams support an XML format phone book download - it would susprice me if the others didn't, but I've no 1st hand experience of them. So you'd need to get some central process to generate the phone books for each type of phone then arrange the phones to download ther own phonebook. Alternatively, use a programmable PBX such as Asterisk to maintain the phone book centrally for you - have it update the name field for incoming calls and allow it to take a short-code for outbound speed-dialling - which can be accessed via a web interface to the PBX in a click to dial sort of thing. Well, that's what I do anyway. It's better than mucking about downloading phone books to all the different types of phones. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, Dec 2, 2010 at 11:19 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. res_phoneprov has had a directory from the start... Try using http://svn.asterisk.org/svn/asterisk/branches/1.8/phoneprov/-directory.xml to get an idea. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
No but if google my posts about IAX2, you will see that I have seen IAX2 cause so many problems with audio, I have made a good amount of money just switching customers to SIP. Even a large ITSP. I have found it to be responsible for poor audio in over a dozen cases and after switching to SIP, the audio was five by. Several people that work for Digium that will remain anonymous, have said to only use IAX when absolutely needed. You will also see people agreeing with me and others that have no issues. I just use SIP. Thanks, Steve T On Thu, Dec 2, 2010 at 9:27 AM, Mark Deneen mden...@gmail.com wrote: Any idea what is it about SIP over IAX2 that made such an improvement? -M On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On 12/02/2010 03:56 PM, Gordon Henderson wrote: On Thu, 2 Dec 2010, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Grandstreams support an XML format phone book download - it would susprice me if the others didn't, but I've no 1st hand experience of them. So you'd need to get some central process to generate the phone books for each type of phone then arrange the phones to download ther own phonebook. Alternatively, use a programmable PBX such as Asterisk to maintain the phone book centrally for you - have it update the name field for incoming calls and allow it to take a short-code for outbound speed-dialling - which can be accessed via a web interface to the PBX in a click to dial sort of thing. Well, that's what I do anyway. It's better than mucking about downloading phone books to all the different types of phones. Gordon Gordon, your idea is not that bad... I will seriously take this into consideration... This is the input I needed :-)... Now to implement it ! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. If you can set up and administer LDAP on the server you will be able to use it on the Snom (and maybe Yealink) phones. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. Here's a good starting point http://www.provu.co.uk/support_snom_ldap.html -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On 12/02/2010 04:33 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. Here's a good starting point http://www.provu.co.uk/support_snom_ldap.html Indeed, found this link also via google. Thank you. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. Does the remote party (being transferred) initially hear hold music, then the line go silent after completing the transfer? No the call just drops and nothing happens in the dial plan. Does the Grandstream show the call still on hold, but you are unable to pick it up? The call just goes a way. Are you using Realtime and/or Direct media? Not using Realtime. I don't think I am using Direct media. Our switch should be handling all of the rtp traffic It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. I have been chasing a deadlock (issue #18403) on blind transfers with 1.8SVN and have not found a work-around yet. While I can deadlock every time (Polycom and Cisco handsets), at least one other has reported different results with the Bria Softphone and Grandstream handsets. You could try a softphone and see if you get the same results as the physical phones. I have a version of Bria I can try later today. -Jonathan Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Hi, Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman: Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alarm POTS lines
Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a fax relay setup locally with hylafax/iaxmodem to solve that issue, and it is working very well. We don't however, have a solution for their alarm lines. The problem is of course that modem calls over VoIP are flaky at best. Even though these alarm calls are low baud rate, when we test with the alarm company we only pass about 30% of the time (ulaw from customer site to our central switch, then out a T1). To be fair there is no QoS on their Internet links yet, and that certainly plays a role. But it seems to me that there should be a solution much like our fax relay, where we literally accept the fax call over the local LAN, produce a PDF file, transfer it to the central switch which then dials it back out over a T1. In that case the only modem over VoIP is on their local LAN, which has performed well for us. I would love to see a DSP modem that could answer an asterisk channel, send the data stream over TCP to some remote asterisk, which could then relay the stream by making an outbound DSP modem call on a PSTN trunk. Has anyone attempted anything like this? As an aside, since the recent thread on Seagate Dockstar installs, I have several running. This would be the perfect platform for the relay on the customer end, being so ridiculously cheap (I bought three for $30 each, plus 3 $10 4G USB sticks). So hoping this will spark some comments on the concept in general, and really hoping someone has actually tackled something similar. It could open up a nice niche for even residential customers with expensive POTS lines dedicated to alarm systems. Cheers, -- Jeff LaCoursiere SunFone j...@sunfone.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] + on Caller-ID
I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 or is it +5705551212 I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] + on Caller-ID
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Sent: Thursday, December 02, 2010 11:13 AM To: asterisk-users Subject: [asterisk-users] + on Caller-ID I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 or is it +5705551212 I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. Just my .02, but 15705551212 (no +) is proper if caller is in U.S., 0115705551212 if not. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] + on Caller-ID
What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 That's the correct one. I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. The country code for the US is 1, which is why +1570... is correct. + means this is an international call and tells a cellphone, for example, to replace + with whatever is the international dialing code for the location where it's currently located. (In the US, that 011, so you'd dial 011-1-570-555-1212, which is the correct way to dial that number.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Karsten I do not see it in the changlog for the 1.8.1 rc version. How would I get the SVN version to test? Thanks for your help. Bryant From: Karsten Wemheuer k...@gmx.de Sent: Thursday, December 02, 2010 11:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issues with 1.8 and BlindTransfer Hi, Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman: Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] + on Caller-ID
On Thu, 2 Dec 2010, Matt wrote: I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 Yes. or is it +5705551212 That would represent a call to Columbia :) I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. The country code for the US is 1 (actually all the NANPA countries, so Canada, Mexico, and much of the Caribbean). 011 is what you dial from within NANPA countries to prefix a country code, so to dial Coumbia, for example, you would dial 011 57 ... From other countries that would be different. From the UK, for example, the same call would be 00 57 xx... (if I recall correctly!). The + represents whatever your local country uses for international access, and only precedes the country code. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 Try the latest SVN branch for 1.8 and see if that resolves your issue: $ svn checkout http://svn.asterisk.org/svn/asterisk/branches/1.8 (that will create a 1.8 directory in your current working directory) On Thu, Dec 2, 2010 at 8:44 AM, Karsten Wemheuer k...@gmx.de wrote: There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 According to the ChangeLog, the fix for issue 18185 was committed after 1.8.1-rc1 was released. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, Dec 2, 2010 at 6:56 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Thu, 2 Dec 2010, Jonas Kellens wrote: I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Grandstreams support an XML format phone book download - it would susprice me if the others didn't, but I've no 1st hand experience of them. Cisco (at least the 79x1 series) phones also have a special XML format for the directory. I have implemented it before as an interactive web app the phones query. No information is stored on the phone itself. Well, that's what I do anyway. It's better than mucking about downloading phone books to all the different types of phones. Real-time query (Live XML/LDAP) back-ended on a database are really the best way to go for Corporate style directory. Unfortunately, you have to get a license from Polycom for LDAP, and static XML files get out of sync way to fast... -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] + on Caller-ID
Some discussion on other lists regarding this, but the + should NOT be part of the sent CLID, and isn't sent by the CLEC's. There IS some discrepancy regarding the 1 in the US. Some send, some do not. This can make for some additional coding when parsing The + is generally used only in print, though some mobile phones add internally. It is unfortunate that there seems to be no solid rule followed. the same can be said regarding dialing in the US, with each state now allowed to set it's own rules. It is generally considered, outside the PUC chambers, that 10 digits for local and 11 digits for toll are proper, but in some locales 11 digits for all calls is mandated, where overlays are in use. Some (diminishing ) locations with 7 digit local dialing still exist. Most expansion ( though not all ) in the last several years have been with NPA overlays. Splits end up as a more expensive solution, especially for the users, and once all electromechanical switches were retired, back in the early 1990's there was no longer a need for splits in NPA's. In summary, no + always a 1 John Novack Jeff LaCoursiere wrote: On Thu, 2 Dec 2010, Matt wrote: I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 Yes. or is it +5705551212 That would represent a call to Columbia :) I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. The country code for the US is 1 (actually all the NANPA countries, so Canada, Mexico, and much of the Caribbean). 011 is what you dial from within NANPA countries to prefix a country code, so to dial Coumbia, for example, you would dial 011 57 ... From other countries that would be different. From the UK, for example, the same call would be 00 57 xx... (if I recall correctly!). The + represents whatever your local country uses for international access, and only precedes the country code. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi on Realtime.
On Thu, 2010-12-02 at 10:52 -0200, Rodrigo Lang wrote: There is no specific Realtime database for chan_dahdi (that I know if). You can store the configuration using Realtime Static using the new chan_dahdi.conf notation without any problems. The only problem with Realtime Static is that you cannot use the text file, you need to load everything from the database. Another possibility would be to use an #exec from chan_dahdi.conf to extract the channel configuration from the database. Thanks for the reply Carlos. You have the model of the tables for chan_dahdi in static mode? This quite difficult to find on the internet ... And you know if the generals can also be included in a static way? Thanks again Here is the table configuration for mysql: CREATE TABLE `ast_config` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) collate utf8_unicode_ci NOT NULL, `category` varchar(128) collate utf8_unicode_ci NOT NULL default 'default', `var_name` varchar(128) collate utf8_unicode_ci NOT NULL, `var_val` varchar(200) collate utf8_unicode_ci NOT NULL, PRIMARY KEY (`id`), KEY `filename_comment` (`filename`,`commented`) ) ENGINE=MyISAM AUTO_INCREMENT=4720 DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci; In extconfig.conf: chan_dahdi.conf = mysql,general,ast_config Database example: +--+++---+-+--+---++ | id | cat_metric | var_metric | commented | filename| category | var_name | var_val| +--+++---+-+--+---++ | 1497 | 27 | 1 | 0 | chan_dahdi.conf | axtel | language | es | | 1498 | 27 | 2 | 0 | chan_dahdi.conf | axtel | context | entrada| | 1499 | 27 | 3 | 0 | chan_dahdi.conf | axtel | usecallerid | yes| | 1500 | 27 | 4 | 0 | chan_dahdi.conf | axtel | hidecallerid | no | | 1501 | 27 | 5 | 0 | chan_dahdi.conf | axtel | callwaiting | no | | 1502 | 27 | 6 | 0 | chan_dahdi.conf | axtel | canpark | no | | 1503 | 27 | 7 | 0 | chan_dahdi.conf | axtel | usecallingpres| yes| | 1504 | 27 | 8 | 0 | chan_dahdi.conf | axtel | callwaitingcallerid | no | | 1505 | 27 | 9 | 0 | chan_dahdi.conf | axtel | threewaycalling | yes| | 1506 | 27 | 10 | 0 | chan_dahdi.conf | axtel | transfer | yes| | 1507 | 27 | 11 | 0 | chan_dahdi.conf | axtel | cancallforward| no | | 1508 | 27 | 12 | 0 | chan_dahdi.conf | axtel | callreturn| yes| | 1509 | 27 | 13 | 0 | chan_dahdi.conf | axtel | echocancel| yes| | 1510 | 27 | 14 | 0 | chan_dahdi.conf | axtel | echocancelwhenbridged | no | | 1511 | 27 | 15 | 0 | chan_dahdi.conf | axtel | echotraining | yes| | 1512 | 27 | 16 | 0 | chan_dahdi.conf | axtel | rxgain| 0.0| | 1513 | 27 | 17 | 0 | chan_dahdi.conf | axtel | txgain| 0.0| | 1514 | 27 | 18 | 0 | chan_dahdi.conf | axtel | busydetect| yes| | 1515 | 27 | 19 | 0 | chan_dahdi.conf | axtel | busycount | 4 | | 1516 | 27 | 20 | 0 | chan_dahdi.conf | axtel | callprogress | no | | 1517 | 27 | 21 | 0 | chan_dahdi.conf | axtel | accountcode | Axtel | | 1518 | 27 | 22 | 0 | chan_dahdi.conf | axtel | amaflags | default| | 1519 | 27 | 23 | 0 | chan_dahdi.conf | axtel | signalling| fxs_ks | | 1520 | 27 | 24 | 0 | chan_dahdi.conf | axtel | group | 1 | | 1521 | 27 | 25 | 0 | chan_dahdi.conf | axtel | faxdetect | incoming | | 1522 | 27 | 26 | 0 | chan_dahdi.conf | axtel | callerid | asreceived | | 1523 | 27 | 27 | 0 | chan_dahdi.conf | axtel | mohinterpret | default| | 1524 | 27 | 28 |
[asterisk-users] DAHDI on VMWARE
Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the REAL machine? Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the “REAL” machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ports
Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk ports Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary Try netstat -anp|grep ast This will show you all of the ports and addresses asterisk is using (if it is running). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Thu, 2 Dec 2010, Gary Kuznitz wrote: Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary You probably see it as: udp0 0 *:sip *:* j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Asterisk ports: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk ports Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary Try netstat -anp|grep ast This will show you all of the ports and addresses asterisk is using (if it is running). Thank you for the reply. Does this look correct? I don't know what port the sip phones are supposed to be communicating on. tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 5382/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 5382/asterisk unix 2 [ ACC ] STREAM LISTENING 180595382/asterisk /var/run/asterisk.ctl unix 2 [ ACC ] STREAM LISTENING 205225768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 2 [ ] DGRAM325885382/asterisk unix 3 [ ] STREAM CONNECTED 207295768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207285768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207275768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205395768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205265768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205255768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205205768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205085768/fast-user-swit Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
On 10-12-02 12:22 PM, Bryant Zimmerman wrote: Karsten I do not see it in the changlog for the 1.8.1 rc version. How would I get the SVN version to test? $ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thank you for the reply. On 2 Dec 2010 at 16:23, Jeff (Jeff LaCoursiere j...@sunfone.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Gary Kuznitz wrote: Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary You probably see it as: udp0 0 *:sip *:* I don't see this. That could certainly be why the phones are connecting. Why wouldn't that port be listening? Thank you, Gary j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk ports On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Asterisk ports: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk ports Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary Try netstat -anp|grep ast This will show you all of the ports and addresses asterisk is using (if it is running). Thank you for the reply. Does this look correct? I don't know what port the sip phones are supposed to be communicating on. tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 5382/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 5382/asterisk snip Thank you, Gary What is the bindport value in sip.conf? The values listed above are 8080 - http 2000 - skinny 5038 - manager 4520 - dundi 4569 - iax I don't have a 2727 on my Asterisk. This might be your sip port. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3s not decoding properly for MusicOnHold.
I have some MP3 files that play well in any MP3 player I throw at them, but when I try to make a MusicOnHold class with them, I get a continuous stream of errors like this: [Dec 2 13:20:31] WARNING[9120]: mp3/common.c:148 decode_header: Layer 2 not supported! [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 50686f74 [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame e7becffc [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443302 I figured this was something that was answered before, but googling for this error message reveals nothing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Behalf Of Gary Kuznitz Shouldn't Asterisk be listening on UDP port 5060? Yes. Unless configured otherwise, that's the SIP port. It's set in sip.conf. What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 unless you know what you're doing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MP3s not decoding properly for MusicOnHold.
On Thu, 2 Dec 2010, Ernie Dunbar wrote: I have some MP3 files that play well in any MP3 player I throw at them, but when I try to make a MusicOnHold class with them, I get a continuous stream of errors like this: Not addressing your errors, but why would you want to burn CPU resources decoding MP3s over and over? If you decode the files to [wav|ulaw|slin|xxx] the files will 'just work' and you'll have more cycles for more fun stuff like handling calls. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, December 02, 2010 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the REAL machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. This is the odd thing - the vmware machine sees the card (dahdi_hardware) but can't seem to properly load via modprobe (conflicts on IRQ 16 (13?) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
On 12/02/2010 04:19 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the REAL machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. This is the odd thing - the vmware machine sees the card (dahdi_hardware) but can't seem to properly load via modprobe (conflicts on IRQ 16 (13?) There is a comment in this post (from 2010-03-14) where someone claims that only certain devices are supported in pass-through: http://communities.vmware.com/message/1493624#1493624 What is the dmesg output when the wctdm24xxp driver is loaded on this system? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, December 02, 2010 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE On 12/02/2010 04:19 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the REAL machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. This is the odd thing - the vmware machine sees the card (dahdi_hardware) but can't seem to properly load via modprobe (conflicts on IRQ 16 (13?) There is a comment in this post (from 2010-03-14) where someone claims that only certain devices are supported in pass-through: http://communities.vmware.com/message/1493624#1493624 What is the dmesg output when the wctdm24xxp driver is loaded on this system? -- Shaun Ruffell [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Thu, Dec 2, 2010 at 4:11 PM, Steve Edwards asterisk@sedwards.com wrote: On Behalf Of Gary Kuznitz Shouldn't Asterisk be listening on UDP port 5060? Yes. Unless configured otherwise, that's the SIP port. It's set in sip.conf. What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress: 0.0.0.0 unless you know what you're doing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Another question to add: Is SIP working, is chan_sip loaded? If your SIP endpoints are not working, try running something simple like sip show peers...if you get a message about no such command existing, SIP is not loading ;-) Cheers, Sherwood McGowan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
There is a comment in this post (from 2010-03-14) where someone claims that only certain devices are supported in pass-through: http://communities.vmware.com/message/1493624#1493624 What is the dmesg output when the wctdm24xxp driver is loaded on this system? [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 Could you update to the lastest version of DAHDI-linux and try: ]# modprobe wctdm24xxp debug=1 and give me the dmesg output? I think the IO registers being in use is a side effect of a previous failed load. I just committed 9503 [1] to bypass that check. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9503 Thanks, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, December 02, 2010 4:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI on VMWARE There is a comment in this post (from 2010-03-14) where someone claims that only certain devices are supported in pass-through: http://communities.vmware.com/message/1493624#1493624 What is the dmesg output when the wctdm24xxp driver is loaded on this system? [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 Could you update to the lastest version of DAHDI-linux and try: ]# modprobe wctdm24xxp debug=1 and give me the dmesg output? I think the IO registers being in use is a side effect of a previous failed load. I just committed 9503 [1] to bypass that check. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9503 Thanks, Shaun [77861.878126] No iBFT detected. [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thanks for the reply. On 2 Dec 2010 at 14:11, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Behalf Of Gary Kuznitz Shouldn't Asterisk be listening on UDP port 5060? Yes. Unless configured otherwise, that's the SIP port. It's set in sip.conf. What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I have re-booted this machine. What else could I look for as to why UDP 5060 isn't listening? Thanks, Gary unless you know what you're doing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alarm POTS lines
On 12/02/2010 10:58 AM, Jeff LaCoursiere wrote: But it seems to me that there should be a solution much like our fax relay, where we literally accept the fax call over the local LAN, produce a PDF file, transfer it to the central switch which then dials it back out over a T1. In that case the only modem over VoIP is on their local LAN, which has performed well for us. It can't be relayed; the alarm protocol is interactive, so a solution analogous to T.38 must be used. I would love to see a DSP modem that could answer an asterisk channel, send the data stream over TCP to some remote asterisk, which could then relay the stream by making an outbound DSP modem call on a PSTN trunk. Has anyone attempted anything like this? Guess what? There is already a standard for this, called V.150, Modem over IP. It certainly would be possible to implement this sort of thing for Asterisk, and some small steps in that direction have been taken way in the past... I'd suggest doing some Google searching to see what you can find. So hoping this will spark some comments on the concept in general, and really hoping someone has actually tackled something similar. It could open up a nice niche for even residential customers with expensive POTS lines dedicated to alarm systems. Or those customers could switch to cell-connected alarm panels (which are rapidly becoming less expensive and provide reliability benefits), or even IP-connected alarm panels. Either choice would be better in the long term than trying to convince an ancient alarm panel's modem to work over a packet network. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Thu, 2 Dec 2010, Steve Edwards wrote: What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 On Thu, 2 Dec 2010, Gary Kuznitz wrote: In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I don't have access to a '1.4' instance right now, but 'sip show settings' works in 1.2 and 1.6 so I'm guessing it should work in 1.4 as well. You may have an error that prevents the SIP channel driver from loading. What do you get with 'unload chan_sip.so' followed by 'load chan_sip.so'? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Version compatibility question...
Hi, Could I install Asterisk 1.4.19, Dahdi 2.4.0 and libpri 1.4.3 ?? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thanks for the reply. On 2 Dec 2010 at 15:57, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Steve Edwards wrote: What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 On Thu, 2 Dec 2010, Gary Kuznitz wrote: In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I don't have access to a '1.4' instance right now, but 'sip show settings' works in 1.2 and 1.6 so I'm guessing it should work in 1.4 as well. You may have an error that prevents the SIP channel driver from loading. What do you get with 'unload chan_sip.so' followed by 'load chan_sip.so'? You get extra points today. I think you found where the problem is. It found /etc/asterisk/sip.conf Warning parse error: No category context for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf. This is what is in sip.conf. [authentication] [general] context = default allowoverlap = no bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes limitonpeers = yes allow = all allowguest=yes What doesn't it like? Thanks, Gary -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astmanproxy on FreeBSD 8.1
Hi. The astmanproxy not work on FreeBSD 8.1 with the Asterisk 1.6.2.13? The ports for the astman was removed. I compiled manually but returns this error when I run voip-freebsd# astmanproxy -dd Dec 2 22:41:11: config: parsing configuration file: /usr/local/etc/asterisk/astmanproxy.conf Dec 2 22:41:11: config: host, localhost,5038,dave,moo,on,off Dec 2 22:41:11: config: retryinterval, 2 Dec 2 22:41:11: config: maxretries, 10 Dec 2 22:41:11: config: sslclienthellotimeout, 200 Dec 2 22:41:11: config: acceptencryptedconnection, yes Dec 2 22:41:11: config: acceptunencryptedconnection, yes Dec 2 22:41:11: config: asteriskwritetimeout, 100 Dec 2 22:41:11: config: clientwritetimeout, 200 Dec 2 22:41:11: config: certfile, /var/lib/asterisk/certs/proxy-server.pem Dec 2 22:41:11: config: listenaddress, * Dec 2 22:41:11: config: listenport, 1234 Dec 2 22:41:11: config: authrequired, no Dec 2 22:41:11: config: proc_user, nobody Dec 2 22:41:11: config: proc_group, nobody Dec 2 22:41:11: config: inputformat, standard Dec 2 22:41:11: config: outputformat, standard Dec 2 22:41:11: config: autofilter, off Dec 2 22:41:11: config: logfile, /var/log/asterisk/astmanproxy.log Dec 2 22:41:11: loading handlers Dec 2 22:41:11: loading: module chan_agent (/usr/local/lib/asterisk/modules/chan_agent.so) dlopen failed: /usr/local/lib/asterisk/modules/chan_agent.so: Undefined symbol option_debug voip-freebsd# Someone have the FreeBSD and the Astmanproxy running? Thanks -- --- Matheus Cucoloto Unix Expertise Voip Expertise -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI on VMWARE
On 12/2/10 4:53 PM, Danny Nicholas wrote: Could you update to the lastest version of DAHDI-linux and try: ]# modprobe wctdm24xxp debug=1 and give me the dmesg output? I think the IO registers being in use is a side effect of a previous failed load. I just committed 9503 [1] to bypass that check. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9503 [77861.878126] No iBFT detected. [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 Are you sure you installed the new drivers? After the new drivers are loaded, what is the output of 'cat /sys/modules/dahdi/version' and 'cat /sys/modules/wctdm24xxp/parameters/debug' ? After a quick scan, I didn't see a code path where -EIO (which is the -5) would be returned from __voicebus_init without another corresponding message in the kernel log. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted
Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. I can't read my voicemails because files seems to be corrupted, for lack of a better word. When I do access my messages, I get those errors: [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3 message(s) in box with max threshold of 100. [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3 message(s) in box with max threshold of 100. [snipped] [Dec 2 19:45:07] WARNING[25993]: app_voicemail.c:7207 play_message: No message attribute file?!! (/var/spool/asterisk/voicemail/xxx/709/INBOX/msg.txt) Well, there isn't a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesn't start at . I have thousands of mailboxes, only one has been reported as having this problem. There might be more though. 1) How do I fix this? I don't mind manually fixing it when it happens, but what's wrong exactly? 2) If this isn't the right list for this (considering it's a SVN question), what is? I'm using SVN because of the blind transfer issue somebody mentioned yesterday. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc/asterisk/sip.conf Warning parse error: No category context
for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf.
This is what is in sip.conf.
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=yes
Running out of clues here :)
I can load the above fine in my 1.2 instance. Any chance the file was
edited on Windows and needs to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] DAHDI on VMWARE
On 12/2/10 6:47 PM, Shaun Ruffell wrote: On 12/2/10 4:53 PM, Danny Nicholas wrote: Could you update to the lastest version of DAHDI-linux and try: ]# modprobe wctdm24xxp debug=1 and give me the dmesg output? I think the IO registers being in use is a side effect of a previous failed load. I just committed 9503 [1] to bypass that check. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9503 [77861.878126] No iBFT detected. [77901.847781] wctdm24xxp :13:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 [77901.847796] wctdm24xxp :13:00.0: IO Registers are in use by another module. [77901.847807] wctdm24xxp :13:00.0: PCI INT A disabled [77901.847816] wctdm24xxp: probe of :13:00.0 failed with error -5 Are you sure you installed the new drivers? After the new drivers are loaded, what is the output of 'cat /sys/modules/dahdi/version' and 'cat /sys/modules/wctdm24xxp/parameters/debug' ? After a quick scan, I didn't see a code path where -EIO (which is the -5) would be returned from __voicebus_init without another corresponding message in the kernel log. Actually I think I wasn't clear. Instead of the latest version of DAHDI-linux I think I should have said to the current trunk of dahdi-linux i.e. ]# svn co http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux-trunk ]# cd dahdi-linux-trunk ]# make install ]# /etc/init.d/dahdi stop ]# dmesg -c /dev/null ]# modprobe wctdm24xxp debug=1 ]# dmesg | tee output.txt The results of output.txt is what I am interested in. Thanks, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc/asterisk/sip.conf Warning parse error: No category context
for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf.
This is what is in sip.conf.
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=yes
Running out of clues here :)
Another thought...
Asterisk can be started with a command line option that specifies the path
to it's configuration file. The default is /etc/asterisk/asterisk.conf.
Does 'ps -aef | grep asterisk' show the '-C' option being used?
Also, Asterisk's configuration file can specify where ('astetcdir') it
should look for sip.conf.
I use these feature to keep each of my client's configuration files in
separate directories, but all on the same development box.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk ports
Thank you very much for the reply.
On 2 Dec 2010 at 17:06, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc/asterisk/sip.conf Warning parse error: No category context
for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf.
This is what is in sip.conf.
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=yes
Running out of clues here :)
I can load the above fine in my 1.2 instance. Any chance the file was
edited on Windows and needs to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
This fixed the problem. There was some garbage in line 1.
You are great. Thank you very much.
Gary
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] + on Caller-ID
When sending CLID in the US it should never contain more than 10 digits (don't include the 1). In fact some providers will BLOCK your call if you do. On Thu, Dec 2, 2010 at 2:24 PM, John Novack jnov...@stromberg-carlson.org wrote: Some discussion on other lists regarding this, but the + should NOT be part of the sent CLID, and isn't sent by the CLEC's. There IS some discrepancy regarding the 1 in the US. Some send, some do not. This can make for some additional coding when parsing The + is generally used only in print, though some mobile phones add internally. It is unfortunate that there seems to be no solid rule followed. the same can be said regarding dialing in the US, with each state now allowed to set it's own rules. It is generally considered, outside the PUC chambers, that 10 digits for local and 11 digits for toll are proper, but in some locales 11 digits for all calls is mandated, where overlays are in use. Some (diminishing ) locations with 7 digit local dialing still exist. Most expansion ( though not all ) in the last several years have been with NPA overlays. Splits end up as a more expensive solution, especially for the users, and once all electromechanical switches were retired, back in the early 1990's there was no longer a need for splits in NPA's. In summary, no + always a 1 John Novack Jeff LaCoursiere wrote: On Thu, 2 Dec 2010, Matt wrote: I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 Yes. or is it +5705551212 That would represent a call to Columbia :) I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback number. The country code for the US is 1 (actually all the NANPA countries, so Canada, Mexico, and much of the Caribbean). 011 is what you dial from within NANPA countries to prefix a country code, so to dial Coumbia, for example, you would dial 011 57 ... From other countries that would be different. From the UK, for example, the same call would be 00 57 xx... (if I recall correctly!). The + represents whatever your local country uses for international access, and only precedes the country code. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted
On Thu, Dec 2, 2010 at 4:56 PM, Mike l...@net-wall.com wrote: Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. There is nothing wrong with using SVN. Well, there isn’t a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesn’t start at . 1) How do I fix this? I don’t mind manually fixing it when it happens, but what’s wrong exactly? I have seen this once on a 1.6.2 system a while back. I just renamed the TXT and audio files to be sequencial numbers starting at and everything worked again. Asterisk assumes the voicemail message files are named that way, and it errors out if that is not the case. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sharing Fail2ban data
Good Day, I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little server that assists in keeping data synchronized across sites. If you're interested in using it to assist in managing your own fail2ban sharing list I'll gladly share it. I also am offering it as a free service for those who are interested in contributing to a blacklist. If you're interested the information is available here: http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in the server code just drop me an email. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
