[Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable
Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 username: anonymous password: your email address -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190, Asterisk and Intercom
I tried to find this in asterisk wiki, but each link I found was broken. How can I use my Snom 190 or 360 softphone as Intercom ? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Re: Cisco 7960 International
actually i do have a 1, i just removed that line because it was setting
callerid
exten = _.,1,SetCallerID(sniped)
--
~Shaun
Tim Robinson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Shaun
I agree with you - I think your dial plan is the problem.
you are stripping off the initial 9 in 'default' thus passing '011xx '
etc to 'outgoing call'.
Outgoing call context needs a 1 in the first priority. the 'n' priority
only seems to work for subsequent steps in the dial plan, as it seems to
stand for 'next priority'. See the wiki on this topic:
http://www.voip-info.org/wiki/index.php?page=Asterisk%20priorities
The outgoing-call line where you have 9011 will not match anything as
you are digit-stripping the leading 9 in 'default' and also needs a 1 in
the priority rather than 'n'
Hope this might help.
Rgds
Tim Robinson
Basingstoke UK
Shaun wrote:
I know it's registering properly because i can use the phone for
internal/local/long distance calls... I suppose it could be a problem in
my dial plan.
I have the following
[default]
exten = _9011.,5,Dial,1,Goto(outgoing-call,${EXTEN:1},1)
[outgoing-call]
exten = _.,n,Dial(SNIPPED)
I also even had this at one point in time...
[outgoing-call]
exten = _9011.,n,Dial(SNIPPED)
Whats weird is that usually when i screw up the dial plan i can see
asterisk via the cli saying somthing, in this case all i have is the sip
debug...
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[Asterisk-Users] cdr_pgsql failing to load in head
I've just upgrade to the latest head (20843) and I get the following error .Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style cdr_pgsql.so (0x0) loaded RTLD_LOCAL Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/cdr_pgsql.so Apr 17 08:41:07 WARNING[8527]: loader.c:753 __load_resource: 5 errors loading module /usr/lib/asterisk/modules/cdr_pgsql.so, aborted Apr 17 08:41:07 WARNING[8527]: loader.c:850 print_and_load: Loading module cdr_pgsql.so failed! Anyone else having this problem. I am running psql 7.4.8 on CENTOS. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP
On 17 Apr 2006, at 00:30, Steve Feinstein wrote: Actually it makes no difference. I tried it in an attempt to get something to happen. Thanks, -Steve Eric ManxPower Wieling wrote: What happens if you remove the r option? r is almost NEVER useful. Steve Feinstein wrote: I've been pulling my hair out over this one trying to understand it. If you have a very simple extension: exten = 1,n,Dial(IAX2/Steve|24|r) Everything I've seen says this should tell the IAX phone (our own iaxclient based one) to make a ringing sound, or asterisk should make the ringback indication itself if it determines that the channel can't do it for itself. But you can dial this extension all day and you never hear a ringback indication. Dial it from a SIP softphone and you do. If you change the default country in the indications.conf, the SIP phone will change the way the ring sounds. IAX, still nothing. You can use PlayTones(ring) in the dialplan before the Dial(), and it seems to behave ok. Playing the appropriate ring indication until the call is answered. But it seems like the behavior is inconsistent with IAX vs. SIP. Is this by design? All the IAX soft phones I've tried are based on the same iaxclient libs, so it's hard to know if it's the phone or asterisk that's not behaving right. Has anyone used an iax hard phone, some other IAX device/software, and does it exhibit the same behavior? Or is this a problem with the iax code not being telling asterisk that IAX phones need to have their indications faked. Any ideas about what's going on would be most gratefully appreciated. I don't know the IAXclient libs, but an IAX client is supposed to send a RINGING packet back after it accepts a call to notify the other end it should generate ringback for the user. The protocol allows it to go straight to ANSWER, or send a PROCEEDING if it hasn't reached the end- point yet. Is your client sending a RINGING packet at the right moment ? Is there a call you should make (after accept but before answer) to get it to send RINGING? Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple asterisk process ?
Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S12:32 0:00 asterisk -vvvg -c asterisk 2520 0.0 5.1 25928 12228 ? S13:21 0:00 asterisk -vvvg -c asterisk 4638 0.0 5.1 25924 12232 ? S13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiple asterisk process ?
I had this and no one could really answer it. I only get it 1 of my systems. I've tried a few things, from removing zaptel watchdog - since I contacted the telco and they said I had a hung channel, to rebuilding * with different options. Are you configuring * manually or using a GUI? Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: 17 April 2006 10:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] multiple asterisk process ? Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S12:32 0:00 asterisk -vvvg -c asterisk 2520 0.0 5.1 25928 12228 ? S13:21 0:00 asterisk -vvvg -c asterisk 4638 0.0 5.1 25924 12232 ? S13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 190, Asterisk and Intercom
Try, http://www.snom.com/wiki/index.php/Asterisk_1.2_Firmware_R4 On 17/04/2006, at 4:29 PM, Ronald Wiplinger wrote: I tried to find this in asterisk wiki, but each link I found was broken. How can I use my Snom 190 or 360 softphone as Intercom ? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Probs with asterisk
Hi all, I am noob with asterisk and i am trying to install it on Debian sarge. I know there is [EMAIL PROTECTED] but i prefere install it on my server wich is yet running an egroupware tool. Phones coulg register the server but when i try to call from one to other (internal call) i get this message : *Verbosity is at least 3* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, 201, 2) exited non-zero on 'SIP/202-b53d'* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/202-b53d'* *-- Unregistered SIP '202'* *-- Registered SIP '202' at 192.168.0.10 port 7737 expires 120 * *an idea someone ? * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
Hi Paul, Thanks for the message! On Sun, 16 Apr 2006, Paul Hewlett wrote: [...] I am curious.. Have you tried disabling CPU1 by setting isolcpus=1 on the kernel command line ? This will make the kernel ignore the second CPU - you can then run asterisk on it by using the taskset command (from schedutils) taskset 0x0001 asterisk -p and asterisk wlll run on a CPU all on its own. I was about to try this and wondered if you might give it a try and report back. I haven't done this yet. Once we have physical access to the machine, I'll make sure we try this out and see what difference it makes. Cheers! Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] te110p and interrupts
I don't know if this only works with multiple cpus but I have HT enabled and it shows cpu0 and cpu1 .. I tried the first part of this email and still the kernel boots and shows 2 cpus.. Will this only work with 2 real cpus? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Begumisa Gerald M |Sent: Monday, April 17, 2006 5:13 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |Hi Paul, | |Thanks for the message! | | On Sun, 16 Apr 2006, Paul Hewlett wrote: | [...] | I am curious.. | | Have you tried disabling CPU1 by setting isolcpus=1 on |the kernel | command line ? | | This will make the kernel ignore the second CPU - you |can then run | asterisk on it by using the taskset command (from schedutils) | | taskset 0x0001 asterisk -p | | and asterisk wlll run on a CPU all on its own. I was about to try | this and wondered if you might give it a try and report back. | |I haven't done this yet. Once we have physical access to the |machine, I'll make sure we try this out and see what |difference it makes. | | |Cheers! |Gerald. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On 4/11/06, Begumisa Gerald M [EMAIL PROTECTED] wrote: Again, if the IO-APIC is reporting that the card is on its own IRQ, it really, truly, honestly *IS* on its own IRQ. The reason that it is suggested to disable the IO-APIC is that on many low-end systems, Allow me to comment that Digium actually recommends turning off APIC and using lspci -vb to troubleshoot this kind of shared-interrupt problem. Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', because that would give everyone a better idea on what to look for when having this kind of problems. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On Mon, 17 Apr 2006, stoffell wrote: Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', [...] Most likely this is why. Regards, Gerald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_pgsql failing to load in head
Chris Stenton wrote: I've just upgrade to the latest head (20843) and I get the following error .Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style cdr_pgsql.so (0x0) loaded RTLD_LOCAL Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/cdr_pgsql.so Apr 17 08:41:07 WARNING[8527]: loader.c:753 __load_resource: 5 errors loading module /usr/lib/asterisk/modules/cdr_pgsql.so, aborted Apr 17 08:41:07 WARNING[8527]: loader.c:850 print_and_load: Loading module cdr_pgsql.so failed! Anyone else having this problem. I am running psql 7.4.8 on CENTOS. As Kevin mentioned early last week, cvs head (or svn trunk) is undergoing major changes that is highly likely to cause it to be more unstable then in the past. Unless you're participating in trunk testing, its probably not the code to use. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. You're trying too hard - unless you tell it not to, the Dial application will do what you're asking. As Olle said, this is the default. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. What is exactly your dial command? bye Ronald Wiplinger Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0615-3, 2006/04/14 Tested on: 2006/4/17 ¤U¤È 07:19:19 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. The canreinvite=yes is required, however your Dial statements used to complete calls between the sip devices cannot use several of the options including t, T, etc. If you remove all options from the Dial statement, restart asterisk, and place a test call, those sip phones that can see each other will auto-negotiate rtp directly between them. If they cannot see each other (eg, nat or firewalls involved), they will not auto-negotiate direct rtp. There is no option for you to specify to forced direct rtp. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Probs with asterisk
Hi Emmanuel! It is very hard to answer such a question without having a dialplan (extensions.conf)or SIP configuration (sip.conf). Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel LAZARO Sent: Montag, 17. April 2006 11:57 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Probs with asterisk Hi all, I am noob with asterisk and i am trying to install it on Debian sarge. I know there is [EMAIL PROTECTED] but i prefere install it on my server wich is yet running an egroupware tool. Phones coulg register the server but when i try to call from one to other (internal call) i get this message : *Verbosity is at least 3* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, 201, 2) exited non-zero on 'SIP/202-b53d'* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/202-b53d'* *-- Unregistered SIP '202'* *-- Registered SIP '202' at 192.168.0.10 port 7737 expires 120 * *an idea someone ? * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
So, is there any other option that prevents that from happening? Something that I might have turned on and makes Dial work trough asterisk? I already even removed asterisk completelyu from system and reinstalled it to be fresh new... still all RTP goes trough Asterisk machine. And the server really can't handle many connections this way. Thanks for the help. Peter Bowyer wrote: On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. You're trying too hard - unless you tell it not to, the Dial application will do what you're asking. As Olle said, this is the default. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Hi Ronald! Please check if the following points are NOT activated. * is not using direct phone to phone RTP streams if: -) either of the clients is configured with canreinvite=no -) the clients cannot agree on a common set of codecs and * needs to perform codec conversion -) either of the clients is configured with nat=yes -) * needs to listen to DTMF tones during the call (for transfers or any other features) Hope this helps, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiago Stein D`Agostini Sent: Montag, 17. April 2006 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP conections,with RTP not going trough Asterisk Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Thanks, that was the problem, I had the t option on the Dial application. Nor that I removed them it works. Thank you. Rich Adamson wrote: Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. The canreinvite=yes is required, however your Dial statements used to complete calls between the sip devices cannot use several of the options including t, T, etc. If you remove all options from the Dial statement, restart asterisk, and place a test call, those sip phones that can see each other will auto-negotiate rtp directly between them. If they cannot see each other (eg, nat or firewalls involved), they will not auto-negotiate direct rtp. There is no option for you to specify to forced direct rtp. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: So, is there any other option that prevents that from happening? Something that I might have turned on and makes Dial work trough asterisk? I already even removed asterisk completelyu from system and reinstalled it to be fresh new... still all RTP goes trough Asterisk machine. And the server really can't handle many connections this way. What options are you using? Post an extract of your dialplan and sip.conf. And how are you determining that the RTP is going through Asterisk? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Probs with asterisk
I had log entries similar to his, bt a reload solved it -- I still wonder what happened. on Monday 04/17/2006 Alex Mosburger([EMAIL PROTECTED]) wrote Hi Emmanuel! It is very hard to answer such a question without having a dialplan (extensions.conf)or SIP configuration (sip.conf). Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel LAZARO Sent: Montag, 17. April 2006 11:57 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Probs with asterisk Hi all, I am noob with asterisk and i am trying to install it on Debian sarge. I know there is [EMAIL PROTECTED] but i prefere install it on my server wich is yet running an egroupware tool. Phones coulg register the server but when i try to call from one to other (internal call) i get this message : *Verbosity is at least 3* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, 201, 2) exited non-zero on 'SIP/202-b53d'* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/202-b53d'* *-- Unregistered SIP '202'* *-- Registered SIP '202' at 192.168.0.10 port 7737 expires 120 * *an idea someone ? * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Remco Barende wrote: So, to document this, the likelihood of a fax working goes in this order best to worse: 1. POTS - fax 2. POTS - FXO-TDM400P-FXS - fax 3. T1 - TE410P - channel bank - fax 4. T1 - TE110P - PCI - TE110P - channel bank - fax 5. T1 - TE110P - PCI - TDM400P-FXS - fax 6. T1 - TE110P - PCI - Ethernet/IP - IAXy - fax 7. FXO-TDM400P - PCI - Ethernet/IP - IAXy - fax Is this a correct? If it's not a PCI problem then there shouldn't be much of a difference between options 3 and 4. If it's a card issue then it would be nice to know which T1 cards handle fax better than others. Yes, BUT!!! be aware that if you have an E1 pri from your telco a T1 channel bank will not help anything. In this case (your option 3) native bridging will be possible and asterisk will have to transcode giving you the some problems again. I don't believe you will ever get POTS - FXO-TDM400P-to-anything to work properly due to TDM card limitations. So, move all of those to the bottom of your list. If you pay close attention to those postings from the last two years in which users say fax works, the majority of them (if not all) are based on either a T1/E1 pstn connection, or, another piece of external hardware that causes fax transmissions to bypass the TDM card. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk redundancy
I'd like to start a discussion about Asterisk redundancy. I know this has been covered in the past, but would like to get an idea of what people are doing for a production system that must be up all the time. Assuming a single E1 out. Here are some of my ideas. HA Linux between the two asterisk boxes. But I am not sure how the Asterisk DB would handle a fail over. What happens to the SIP registrations? Can the Asterisk DB be offloaded to MYSQL for example? The local DB is importatn because this is a call center with agents logged in to multiple queues. Config could either be realtime or duplicated manually. What about recorded message, has anyone had any problems with an NFS volume providing recorded messages such as periodic messages in queues? This solution would require a manual swap of the E1 cable inthe event of failure. Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW that would allow failover to an alternate Asterisk box without manually switching the cable? This one is a litte expensive(http://www.mapleleaf-technologies.com/webstore/ethernetbridges.php ), but seems like it would do the trick. But I would have to run TDMoE between the Asterisk boxes and the bridge. Not a big deal probably, but I have no experience with TDMoE. I would appreciate any comments regarding redundancy, and how people are solving these problems. Regards to all, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 4/17/06, Alex Mosburger [EMAIL PROTECTED] wrote: -) * needs to listen to DTMF tones during the call (for transfers or any other features) Does this mean you cannot do any blind or attended transfer? or only the # transfer option (asterisk built-in, from features.conf) doesn't work? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
On Monday 17 April 2006 07:44, Rich Adamson wrote: I don't believe you will ever get POTS - FXO-TDM400P-to-anything to work properly due to TDM card limitations. So, move all of those to the bottom of your list. I *had* this working. POTS - TDM400 TDM400 - Real_honest_fax_machine As I'd posted several times already. I have not been able to repeat this success, though. If you pay close attention to those postings from the last two years in which users say fax works, the majority of them (if not all) are based on either a T1/E1 pstn connection, or, another piece of external hardware that causes fax transmissions to bypass the TDM card. Correct. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Probs with asterisk
i tried reload but nothing ;) maybe prob with my dialplan like says alex, i'll try to solve it remaking my dialplan. John covici a écrit : I had log entries similar to his, bt a reload solved it -- I still wonder what happened. on Monday 04/17/2006 Alex Mosburger([EMAIL PROTECTED]) wrote Hi Emmanuel! It is very hard to answer such a question without having a dialplan (extensions.conf)or SIP configuration (sip.conf). Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel LAZARO Sent: Montag, 17. April 2006 11:57 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Probs with asterisk Hi all, I am noob with asterisk and i am trying to install it on Debian sarge. I know there is [EMAIL PROTECTED] but i prefere install it on my server wich is yet running an egroupware tool. Phones coulg register the server but when i try to call from one to other (internal call) i get this message : *Verbosity is at least 3* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, 201, 2) exited non-zero on 'SIP/202-b53d'* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/202-b53d'* *-- Unregistered SIP '202'* *-- Registered SIP '202' at 192.168.0.10 port 7737 expires 120 * *an idea someone ? * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Probs with asterisk
hmm In sip.conf i have the declaration for 2 phones (i am testing asterisk installation). No prob with phones identification (i think). I will take a look at my extension.conf if i hadn't make a mistake (i don't really understood how it works yet). Alex Mosburger a écrit : Hi Emmanuel! It is very hard to answer such a question without having a dialplan (extensions.conf)or SIP configuration (sip.conf). Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel LAZARO Sent: Montag, 17. April 2006 11:57 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Probs with asterisk Hi all, I am noob with asterisk and i am trying to install it on Debian sarge. I know there is [EMAIL PROTECTED] but i prefere install it on my server wich is yet running an egroupware tool. Phones coulg register the server but when i try to call from one to other (internal call) i get this message : *Verbosity is at least 3* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, 201, 2) exited non-zero on 'SIP/202-b53d'* *-- Executing Set(SIP/202-b53d, TIMEOUT(absolute)=15) in new stack* *-- Channel will hangup at 2006-04-16 20:36:05 UTC.* *-- Executing Congestion(SIP/202-b53d, ) in new stack* *== Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/202-b53d'* *-- Unregistered SIP '202'* *-- Registered SIP '202' at 192.168.0.10 port 7737 expires 120 * *an idea someone ? * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Rich Adamson wrote: I don't believe you will ever get POTS - FXO-TDM400P-to-anything to work properly due to TDM card limitations. So, move all of those to the bottom of your list. If you pay close attention to those postings from the last two years in which users say fax works, the majority of them (if not all) are based on either a T1/E1 pstn connection, or, another piece of external hardware that causes fax transmissions to bypass the TDM card. I and other iaxmodem users can say fax works with analog PSTN connections. In my case, as well as those others of which I am aware, an X100P (clone, er winmodem) is being used. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk redundancy
As a matter of curiosity, does anyone know what the E1/T1 interface in this (redfone) box is ? Could the box be an embedded linux device with a PCI slot, running linux and therefore zaptel, and therefore the PCI card could be a Digium or sangoma card ... Any clues ? Does anyone have such a box they could dismantle ;) Julian. Joseph Rothstein wrote: I'd like to start a discussion about Asterisk redundancy. I know this has been covered in the past, but would like to get an idea of what people are doing for a production system that must be up all the time. Assuming a single E1 out. Here are some of my ideas. HA Linux between the two asterisk boxes. But I am not sure how the Asterisk DB would handle a fail over. What happens to the SIP registrations? Can the Asterisk DB be offloaded to MYSQL for example? The local DB is importatn because this is a call center with agents logged in to multiple queues. Config could either be realtime or duplicated manually. What about recorded message, has anyone had any problems with an NFS volume providing recorded messages such as periodic messages in queues? This solution would require a manual swap of the E1 cable inthe event of failure. Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW that would allow failover to an alternate Asterisk box without manually switching the cable? This one is a litte expensive(http://www.mapleleaf-technologies.com/webstore/ethernetbridges.php ), but seems like it would do the trick. But I would have to run TDMoE between the Asterisk boxes and the bridge. Not a big deal probably, but I have no experience with TDMoE. I would appreciate any comments regarding redundancy, and how people are solving these problems. Regards to all, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk redundancy
On 17 Apr 2006, at 12:58, Joseph Rothstein wrote: I'd like to start a discussion about Asterisk redundancy. I know this has been covered in the past, but would like to get an idea of what people are doing for a production system that must be up all the time. Assuming a single E1 out. Here are some of my ideas. HA Linux between the two asterisk boxes. But I am not sure how the Asterisk DB would handle a fail over. What happens to the SIP registrations? Can the Asterisk DB be offloaded to MYSQL for example? The local DB is importatn because this is a call center with agents logged in to multiple queues. Config could either be realtime or duplicated manually. What about recorded message, has anyone had any problems with an NFS volume providing recorded messages such as periodic messages in queues? This solution would require a manual swap of the E1 cable inthe event of failure. Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW that would allow failover to an alternate Asterisk box without manually switching the cable? This one is a litte expensive(http://www.mapleleaf-technologies.com/webstore/ ethernetbridges.php ), but seems like it would do the trick. But I would have to run TDMoE between the Asterisk boxes and the bridge. Not a big deal probably, but I have no experience with TDMoE. I would appreciate any comments regarding redundancy, and how people are solving these problems. Regards to all, Joe I strongly advise you to get the economics clear before you proceed. Get an estimate of the business costs of (say) 1hour's downtime every 3 years. Once you have done that you have a budget to work to. If you don't and you just follow the No downtime ever! rule two things will happen: 1) you will spend a boatload of money, probably far more than needed. 2) You will fail. 100% uptime doesn't happen - ever - folks get close, but every step costs exponentially more, and gets exponentially more complex - so much harder to maintain - so more fragile. Also don't forget to talk to your telco and hear what they can do for you. You may find that they can detect a PRI failure and move calls to a fallback number (mobiles, analog, answerphone, voicemail,voip provider). I _very_ much doubt you can find a solution where you can have the box at your end of the PRI fail (be it an asterisk or a switching box) and still keep the current calls. Keep us posted on what you find, it is an important topic and my views (keep it simple) aren't typical :-) Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
On Monday 17 April 2006 08:21, Lee Howard wrote: I and other iaxmodem users can say fax works with analog PSTN connections. In my case, as well as those others of which I am aware, an X100P (clone, er winmodem) is being used. Interesting. Do you have more information about your setup (asterisk and zaptel versions, iaxmodem version, configuration for each, etc.)? I wouldn't mind trying iaxmodem on FXS with the TDM400... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Lee Howard wrote: Rich Adamson wrote: I don't believe you will ever get POTS - FXO-TDM400P-to-anything to work properly due to TDM card limitations. So, move all of those to the bottom of your list. If you pay close attention to those postings from the last two years in which users say fax works, the majority of them (if not all) are based on either a T1/E1 pstn connection, or, another piece of external hardware that causes fax transmissions to bypass the TDM card. I and other iaxmodem users can say fax works with analog PSTN connections. In my case, as well as those others of which I am aware, an X100P (clone, er winmodem) is being used. Lee. I agree with Lee. I have about 30 machines in production using iaxmodem and hylafax which work perfectly. Most are running off of T1s, but some are on TDM400 and TDM2400s. I only use IBM servers (which are about twice the cost for the low end Dells), and have never had to resolve an IRQ problem. I just looked up the hylafax usage reports on those people running the analog FXOs, and one of them had 390 pages in the last week, only one error, which I would consider acceptable. Thanks, Nick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CentOS 4.x Asterisk RPMS
Andrew, the only two patches are the ones you mention here?(spandsp and iaxmodem)? no other patches? Thanks, On 4/15/06, alist [EMAIL PROTECTED] wrote: I have complied the latest releases, patched with spandsp and iaxmodem support. If you upgrade to the provided kernel you will have support for the zaptel modules and sangoma drivers without the need to recomplie anything. ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS4/ Have fun... Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Andrew Kohlsmith wrote: On Monday 17 April 2006 08:21, Lee Howard wrote: I and other iaxmodem users can say fax works with analog PSTN connections. In my case, as well as those others of which I am aware, an X100P (clone, er winmodem) is being used. Interesting. Do you have more information about your setup (asterisk and zaptel versions, iaxmodem version, configuration for each, etc.)? Currently *I'm* using Asterisk 1.2.3 and zaptel 1.2.2, but others are using many variety of versions, both older and newer. I've used every version of iaxmodem. :-) I would be happy to share my configuration files with everyone, but all of it is really quite bland... mostly default configs for everything. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR query
Alex Brett wrote: The problem I have, is that the 'billsec' field in the CDR records, only starts ticking if I accept the call, so it isn't including the time that I have answered the call on my mobile, but not actually accepted the call, which means if I reject the call or whatever, then it doesn't log the fact that I have actually spent money making a PSTN phone call... What I guess I'm looking for is some sort of command to put into my macro to start the 'billsec' counting, does anybody know if such a thing exists, or if there is some other way of getting round this problem? (I could just use the 'duration' field, but this would include time where it was merely ringing and I'm not actually being charged etc). For anybody interested, I got round this by hacking at app_dial.c and adding an extra flag (b) that will set the CDR record as answered as soon as it starts the macro. Now billsec is accurate, only downside being that it shows all the calls as ANSWERED even if the callee rejected it or whatever - I'm looking at finding a way round that, then if I can make it nice enough I'll submit a patch etc - contact me off-list if you want it in it's present state... Alex -- Alex Brett [EMAIL PROTECTED] http://www.loho.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents, Queues, and Voicemail
All,I am experiencing an issue where if an agent is logged into the queue, but has their client closed. It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable. This pulls the call out of the queue, which is not what I would like to happen. I am wondering if this is the expected behavior and I should rewrite my macro to handle checking if an agent is logged into the queue, or if there is a way to realize when a call comes from the Queue() application to not dump to voicemail and just ring the agent. My ideal setup would be agents can log into a queue. If a call comes from the Queue() application it get's passed around via round robin (or whatever I have configured). If the call does not get passed from Queue() (a normal call), then proceed to the user's macro and go to voicemail, etc.. Please let me know if you have questions about my setup.Thanks,Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
Asterisk is a multithreaded system. I have not in mind how many threads open and where. But ie, if you have enabled pbx_spool.so to generate calls from files, that module launch its own thread to monitor the calls directory, MOH launch other thread, every channel has its own thread, the CLI has its own thread, to listen for calls in SIP, IAX, you need specific threads etc, etc, i would say what you see is normal. Regards On 4/17/06, Lee Archer [EMAIL PROTECTED] wrote: I had this and no one could really answer it. I only get it 1 of my systems. I've tried a few things, from removing zaptel watchdog - since I contacted the telco and they said I had a hung channel, to rebuilding * with different options. Are you configuring * manually or using a GUI? Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: 17 April 2006 10:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] multiple asterisk process ? Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S12:32 0:00 asterisk -vvvg -c asterisk 2520 0.0 5.1 25928 12228 ? S13:21 0:00 asterisk -vvvg -c asterisk 4638 0.0 5.1 25924 12232 ? S13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall and Fax
Carlos, Make sure you are using the telco clock in your zaptel.conf, and use faxdetect=both in your unicall.conf. I had several issues before finding a combination of asterisk+libspandsp+libunicall that worked for me. Currently they are: - spandsp-20060205.tar.gz - libmfcr2-20060205.tar.gz - asterisk-1.2.4.tar.gz Saludos, -- Paulo Carlos Chavez escreveu: On Fri, 14 Apr 2006 23:48:31 -0300, Paulo Scardine wrote I use Unicall and app_rxfax to receive (works very well) and iaxmodem+hylafax for sending (took a little longer than I would expect, like 50%). Just use a recent iaxmodem (make it static). I use a libspandsp snapshot. Did you do anything special in the configuration? I hae tried several things but I simply cannot send or receive a fax when using Unicall. Faxes on the same machine using a couple of FXO channels on a TDM400P card work most of the time although they usually cause the sending machine to print an error on the transmission. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agents, Queues, and Voicemail
How do you use the agents? Callback or on-hook? If callback you can direct the calls to another context that doesn't have the fail over to voicemail. --johann Kyle Sexton wrote: All, I am experiencing an issue where if an agent is logged into the queue, but has their client closed. It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable. This pulls the call out of the queue, which is not what I would like to happen. I am wondering if this is the expected behavior and I should rewrite my macro to handle checking if an agent is logged into the queue, or if there is a way to realize when a call comes from the Queue() application to not dump to voicemail and just ring the agent. My ideal setup would be agents can log into a queue. If a call comes from the Queue() application it get's passed around via round robin (or whatever I have configured). If the call does not get passed from Queue() (a normal call), then proceed to the user's macro and go to voicemail, etc.. Please let me know if you have questions about my setup. Thanks, Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
Polycom IP501 Assembly: 2345-11500-040 Rev: B Bootrom: 3.1.0.0269 SIP Ver: 1.6.5.0043 I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press Send. Nothing appears on the Asterisk CLI screen. This phone worked briefly before upgrading from br:2.6.1 and sip:1.6.2 I've heard that I can't go back to bootrom versions 2.x once having upgraded it to 3.x. Any clues where I should begin my search for an answer? Thanks! Jim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agents, Queues, and Voicemail
Johann,I'm using callback for the login method. That definitely makes sense, I'll try it out and see. Thanks!Kyle SextonOn 4/17/06, Johann [EMAIL PROTECTED] wrote: How do you use the agents?Callback or on-hook?If callback you can direct thecalls to another context that doesn't have the fail over to voicemail.--johannKyle Sexton wrote: All, I am experiencing an issue where if an agent is logged into the queue, but has their client closed.It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable.This pulls the call out of the queue, which is not what I would like to happen.I am wondering if this is the expected behavior and I should rewrite my macro to handle checking if an agent is logged into the queue, or if there is a way to realize when a call comes from the Queue() application to not dump to voicemail and just ring the agent. My ideal setup would be agents can log into a queue.If a call comes from the Queue() application it get's passed around via round robin (or whatever I have configured).If the call does not get passed from Queue() (a normal call), then proceed to the user's macro and go to voicemail, etc.. Please let me know if you have questions about my setup. Thanks, Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk voicemail question
Tofik Suleymanov wrote: When new voicemail comes and i pick up the phone i hear special tones indicating that the new voicemail arrived. 2. there is no new voicemail (checked mailbox on filesystem), but when i pick up the phone i hear speial tones indicating that there is a new message We had a case of this here recently. It turns out that the msg0002.txt file in the Inbox of one of the mailboxes did not get deleted. (asterisk 1.2.6) Because of this file, Asterisk kept the message waiting light lit. However, going through voicemail did not allow the deleting of file. Check to see if you have any erroneous files in the /var/spool/asterisk/voicemail/default/xxx/INBOX directory. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk redundancy
I'd like to start a discussion about Asterisk redundancy. I know this has been covered in the past, but would like to get an idea of what people are doing for a production system that must be up all the time. I'm going to pipe in on this one. Asterisk redundancy is a huge discussion on this list at LEAST once a month, so you would think we would have all the details all hammered out and on the wiki and say Hey, we had that discussion last week... what we came up with is on the wiki now. HA Linux between the two asterisk boxes. But I am not sure how the Asterisk DB would handle a fail over. What happens to the SIP registrations? Can the Asterisk DB be offloaded to MYSQL for example? The local DB is importatn because this is a call center with agents logged in to multiple queues. Config could either be realtime or duplicated manually. What about recorded message, has anyone had any problems with an NFS volume providing recorded messages such as periodic messages in queues? This solution would require a manual swap of the E1 cable inthe event of failure. Not sure how the agents would work out, you may need a server that handles the queueing setup. We've got 5 servers here (one voicemail, two call servers, and two gateways). The way it's configured, any DB information is replicated through a series of dialplan magic and scripts, so I'm not sure what it would require to replicate agent information. I will tell you straight up that NFS mounted volumes will cause asterisk to croak if it needs access to something that's not mounted. The first time the NFS share disappears for a moment, you're going to be restarting services and losing time on the asterisk machines that need the mounts. It would be better to drop the files on all the systems so you don't have to worry about that. Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW that would allow failover to an alternate Asterisk box without manually switching the cable? This one is a litte expensive(http://www.mapleleaf-technologies.com/webstore/ethernetbridges.php ), but seems like it would do the trick. But I would have to run TDMoE between the Asterisk boxes and the bridge. Not a big deal probably, but I have no experience with TDMoE. My experience of TDMoE is limited, all of our servers talk sip when a call is moved from one box to another. We have double T1 lines per gateway server, so if one goes down, we move to half capacity. Any calls in progress on the one gateway die, but everything else automatically moves to using only the other gateway. I would appreciate any comments regarding redundancy, and how people are solving these problems. Regards to all, Joe -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Monday 17 April 2006 10:17, Jim Rice wrote: I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press Send. a sip debug on the asterisk console will give you a ton of data if it's getting to the asterisk box... I guess that's your first step, to see if the phone is trying to do some kind of screwy routing or not? (sip no debug is the command to turn it off after you're sick of seeing all the scrolling messages.) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Don't see my post
Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI Installation Eicon Diva Server
On Wed, 12 Apr 2006 [EMAIL PROTECTED] wrote: Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available, but my mistake may be in configuring the trunks... The configuration is as easy as with BRI lines. Can you provide more (like your confs and verbose/debug output)? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Don't see my post
John Rich a écrit : Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. This mailing list is not moderated (as far as I can tell) _but_ your message would be more appropriate on the -biz list anyways. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quick question
Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! -- Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick question
You could try chan_oh323.so and chan_h323.so. I think also ooh323 supports inband DTMFs. Regards Alberto Sagredo Tomislav Parčina escribió: Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! -- Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't see my post
We have the same problem! And our question was a technical one about Snom 360... Was the mailing list server down? I'll go search in my sent items and try to send it again... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Jean-Michel Hiver Verzonden: maandag 17 april 2006 17:04 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Don't see my post John Rich a écrit : Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. This mailing list is not moderated (as far as I can tell) _but_ your message would be more appropriate on the -biz list anyways. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press Send. Nothing appears on the Asterisk CLI screen. Did the upgrade modify the dialplan setting on your phone? This sounds suspiciously like trying to dial a number that is not matched or allowed by the dialplan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Don't see my post
Thanks Do you have any suggestion on which news group I should target? Thanks John,Jean-Michel Hiver [EMAIL PROTECTED] wrote: John Rich a écrit : Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes.This mailing list is not moderated (as far as I can tell) _but_ your message would be more appropriate on the -biz list anyways.Cheers,Jean-Michel.-- Jean-Michel Hiver - http://ykoz.net/Découvrez la Réunion des Technologies IP TelecomTEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries for just 2¢/min with Yahoo! Messenger with Voice.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX phone hardware recommendation
Hi, Can someone plesae recommend a good IAX hard phone? And/or IAX ATA? Thanks John Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1/min.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept garbling my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting CDR dnid and Billing
I need to manually set certain CDR fields. 1). Callers are allowed to call someone within the same organisation by using a 4 digit extension. A database lookup maps the 4 digit extension to the real number. However, a CDR for this call shows the original 4 digit extension still. What variable is used to set the 'dst' field in the CDR? Is it the dnid AGI var? How can I set this? I tried setting the agi variable but it had no effect. 2). I am implementing findme/followme. When someone directs their calls to a new number, we need to set the originator of the call to this person's number. I therefore need to set the 'src' field in the CDR to the number of the person doing the forwarding, not the originator of the call. Is the 'src' field based on the caller id? I sure hope not, because we are letting users set their caller id to something else. How can I set the 'src' field for a CDR? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Mon, 2006-04-17 at 10:45 -0400, Andrew Kohlsmith wrote: a sip debug on the asterisk console will give you a ton of data if it's getting to the asterisk box... *CLI sip debug ... stuff scrolled off screen ... m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.201 : 5060 (non-NAT) Found user '201jim' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.201:2230 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw| alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 6600546 in office (domain 10.0.0.1) Reliably Transmitting (no NAT) to 10.0.0.201:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.0.0.201;branch=z9hG4bKfaa55b2b1A2706FC;received=10.0.0.201 From: 201 JIM sip:[EMAIL PROTECTED];tag=5B7B9FBA-441AF0A9 To: sip:[EMAIL PROTECTED];user=phone;tag=as7bb06f68 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- SIP read from 10.0.0.201:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.201;branch=z9hG4bKfaa55b2b1A2706FC From: 201 JIM sip:[EMAIL PROTECTED];tag=5B7B9FBA-441AF0A9 To: sip:[EMAIL PROTECTED];user=phone;tag=as7bb06f68 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- Jim Rice by Design Publishing 11626 N. Tracey Road Hayden, Idaho 83835 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
I'm sorry to hear that, but I didn't experience such a problem, 6.0 seems to work quite well on my phone. Do you have a suggestion for my question? Or alternative: Is it possible to send a custom SIP NOTIFY message (with XML body) to an asterisk sip client? - Frederic -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Dr. Michael J. Chudobiak Verzonden: maandag 17 april 2006 18:45 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept garbling my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
On Monday 17 April 2006 16:03, Moises Silva wrote: Asterisk is a multithreaded system. I have not in mind how many threads open and where. But ie, if you have enabled pbx_spool.so to generate calls from files, that module launch its own thread to monitor the calls directory, MOH launch other thread, every channel has its own thread, the CLI has its own thread, to listen for calls in SIP, IAX, you need specific threads etc, etc, i would say what you see is normal. Regards This is incorrect. Asterisk is a multithreaded system but how the threads are handled by the OS depends on the version of threads that is being used. For Linuxthreads (kernel 2.4), one would see a separate entry for each thread when executing 'ps aux'. For NPTL (linux 2.6) one does not see each thread as a separate entry. So the OP must tell us which kernel version he is using. Alternatively type getconf GNU_LIBPTHREAD_VERSION as root. For NPTL u should get something like NPTL 2.3.5 or suchlike. If you are using NPTL and there is more than one entry for asterisk, then asterisk has spawned an extra process for some reason. If extra processes keep appearing then I would say that he has a bug or error somewhere and asterisk is respawning that separate process. Paul Hewlett On 4/17/06, Lee Archer [EMAIL PROTECTED] wrote: I had this and no one could really answer it. I only get it 1 of my systems. I've tried a few things, from removing zaptel watchdog - since I contacted the telco and they said I had a hung channel, to rebuilding * with different options. Are you configuring * manually or using a GUI? Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: 17 April 2006 10:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] multiple asterisk process ? Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S12:32 0:00 asterisk -vvvg -c asterisk 2520 0.0 5.1 25928 12228 ? S13:21 0:00 asterisk -vvvg -c asterisk 4638 0.0 5.1 25924 12232 ? S13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_pgsql failing to load in head
Rich Adamson wrote: Chris Stenton wrote: I've just upgrade to the latest head (20843) and I get the following error .Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style cdr_pgsql.so (0x0) loaded RTLD_LOCAL Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/cdr_pgsql.so Apr 17 08:41:07 WARNING[8527]: loader.c:753 __load_resource: 5 errors loading module /usr/lib/asterisk/modules/cdr_pgsql.so, aborted Apr 17 08:41:07 WARNING[8527]: loader.c:850 print_and_load: Loading module cdr_pgsql.so failed! Anyone else having this problem. I am running psql 7.4.8 on CENTOS. As Kevin mentioned early last week, cvs head (or svn trunk) is undergoing major changes that is highly likely to cause it to be more unstable then in the past. Unless you're participating in trunk testing, its probably not the code to use. Those are the messages kicked out by the module loader, signifying that the module in question has not yet been upgraded to be compatible with it. AFAIK that one is now upgraded, and I see the Postgres CDR module got fixed this morning. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Billing Server Open Source
Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On Monday 17 April 2006 12:39, Anton Krall wrote: I don't know if this only works with multiple cpus but I have HT enabled and it shows cpu0 and cpu1 .. I tried the first part of this email and still the kernel boots and shows 2 cpus.. Will this only work with 2 real cpus? I believe so. Hyperthreading is not really SMP. Additionally I think that HT should not be used if asterisk is doing a lot of transcoding - whilst HT gives u 2 'pipes' there is still only one FPU and heavy transcoding will simply bottleneck at the FPU(NB SSE/MMX count as FP since they use the FP registers.). If u turn HT on, a 1% penalty is incurred fir the mutexes required so heavy use of the FPU will actually show a decrease in speed compared to no HT. In theory at least. :-) Paul Hewlett |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Begumisa Gerald M |Sent: Monday, April 17, 2006 5:13 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] te110p and interrupts | |Hi Paul, | |Thanks for the message! | | On Sun, 16 Apr 2006, Paul Hewlett wrote: | [...] | | I am curious.. | | Have you tried disabling CPU1 by setting isolcpus=1 on | |the kernel | | command line ? | | This will make the kernel ignore the second CPU - you | |can then run | | asterisk on it by using the taskset command (from schedutils) | | taskset 0x0001 asterisk -p | | and asterisk wlll run on a CPU all on its own. I was about to try | this and wondered if you might give it a try and report back. | |I haven't done this yet. Once we have physical access to the |machine, I'll make sure we try this out and see what |difference it makes. | | |Cheers! |Gerald. |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Like Phone Switch ?
Hi All, I have a project about IP telephony, we want to build a switch phone using Asterisk, the amount of users is of 15000 and also it was needed to connect with other PSTN using SS7 my consult is: asterisk can be used for this ammount?, Can I to build a switch and using SS7 for example the SS7 of teleprime www.telerpime.com (SS7 to SIP) Thanks in advanced, Regards _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
On Mon, 2006-04-17 at 19:12 +0200, Paul Hewlett wrote: This is incorrect. Asterisk is a multithreaded system but how the threads are handled by the OS depends on the version of threads that is being used. For Linuxthreads (kernel 2.4), one would see a separate entry for each thread when executing 'ps aux'. For NPTL (linux 2.6) one does not see each thread as a separate entry. So the OP must tell us which kernel version he is using. Alternatively type getconf GNU_LIBPTHREAD_VERSION as root. For NPTL u should get something like NPTL 2.3.5 or suchlike. If you are using NPTL and there is more than one entry for asterisk, then asterisk has spawned an extra process for some reason. If extra processes keep appearing then I would say that he has a bug or error somewhere and asterisk is respawning that separate process. Are you sure? root 2532 0.0 0.2 2532 620 ?S17:22 0:00 /bin/sh /usr/sbin/safe_asterisk root 2539 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2542 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2544 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2545 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2546 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2547 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2548 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2549 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2550 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2551 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2552 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2553 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2554 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2555 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2556 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2557 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2558 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2559 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2560 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2561 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2562 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2563 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2564 0.0 2.8 17716 7316 ?S17:22 0:01 asterisk -n -vvvg -c root 2565 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2566 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2567 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c With NPTL 2.3.6 If that is the case * is totally hosed, no? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Monday 17 April 2006 12:53, Jim Rice wrote: Looking for 6600546 in office (domain 10.0.0.1) Reliably Transmitting (no NAT) to 10.0.0.201:5060: SIP/2.0 484 Address Incomplete It looks like whatever you're dialing from the IP501, Asterisk doesn't like. Did the number you called (6600546) have a match in the dialplan context it's entering in to (office)? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording
Hi Waldo, The best I've seen so far is about 100 concurrent calls on a single Xeon 2.4Ghz. The CPU was 100% but this does not mean anything since this is due to GSM encoding which happens sequentially and always leaves capture work in priority. I'm sure it can do more than that, it's just not been tested so far. Cheers Henri On 12/04/06, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hey Henri, Long time no talk. How far have you been able to scale oreka up to? How many simultaneous calls have you been able to record and under what hardware config? Thanks, Waldo On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote: Another solution would be to use a dedicated recording server sniffing RTP and signalling packets in the media path using software such as http://www.oreka.org. Oreka automatically mixes both legs of an RTP conversation to disk and GSM encodes the result in a separate thread so that capture always has priority. Cheers Henri ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Notify cisco-check-cfg - Does it still work with 8.2?
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg doesn't elicit any response from the phone using fw 8.2? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
Thanks for clarifying that Paul. my output for getconf is: linuxthreads-0.10 so i guess is normal to have several threads shown by ps axu right? On 4/17/06, Dave Cotton [EMAIL PROTECTED] wrote: On Mon, 2006-04-17 at 19:12 +0200, Paul Hewlett wrote: This is incorrect. Asterisk is a multithreaded system but how the threads are handled by the OS depends on the version of threads that is being used. For Linuxthreads (kernel 2.4), one would see a separate entry for each thread when executing 'ps aux'. For NPTL (linux 2.6) one does not see each thread as a separate entry. So the OP must tell us which kernel version he is using. Alternatively type getconf GNU_LIBPTHREAD_VERSION as root. For NPTL u should get something like NPTL 2.3.5 or suchlike. If you are using NPTL and there is more than one entry for asterisk, then asterisk has spawned an extra process for some reason. If extra processes keep appearing then I would say that he has a bug or error somewhere and asterisk is respawning that separate process. Are you sure? root 2532 0.0 0.2 2532 620 ?S17:22 0:00 /bin/sh /usr/sbin/safe_asterisk root 2539 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2542 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2544 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2545 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2546 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2547 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2548 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2549 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2550 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2551 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2552 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2553 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2554 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2555 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2556 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2557 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2558 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2559 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2560 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2561 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2562 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2563 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2564 0.0 2.8 17716 7316 ?S17:22 0:01 asterisk -n -vvvg -c root 2565 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2566 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2567 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c With NPTL 2.3.6 If that is the case * is totally hosed, no? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
In Cisco land you would send a command to the phone via a long URL so the idea was to send a HTTP/POST to ipaddress/ciscoservices/directory/browser/index.html?url=http://somedomain/services/app.php this is not exact as it changes often. I am reading on the snoms I am sure their system is much better than Cisco's in the openess department :) On 4/17/06, TWV [EMAIL PROTECTED] wrote: I'm sorry to hear that, but I didn't experience such a problem, 6.0 seems to work quite well on my phone. Do you have a suggestion for my question? Or alternative: Is it possible to send a custom SIP NOTIFY message (with XML body) to an asterisk sip client? - Frederic -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Dr. Michael J. Chudobiak Verzonden: maandag 17 april 2006 18:45 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept garbling my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Billing Server Open Source
http://www.asterisk2billing.org/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 17, 2006 1:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Billing Server Open Source Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance. __ NOD32 1.1492 (20060416) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RES: [Asterisk-Users] attended transfer issue
dovb wrote: That fix would be great!!! To press # and be able to get the call back and terminate the transfer... I had to implement an horrible workaround to emulate this functionality Well, as it stands now, to hangup while you are doing a transfer, you using the hangup feature code (in features.conf). That will put you back to the original person. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Billing Server Open Source
FYI, this is more of a question for the asterisk-biz list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 17, 2006 1:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Billing Server Open Source Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance. __ NOD32 1.1492 (20060416) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Mon, 2006-04-17 at 11:48 -0400, Jonathan k. Creasy wrote: Did the upgrade modify the dialplan setting on your phone? This sounds suspiciously like trying to dial a number that is not matched or allowed by the dialplan. I'm not sure. I think it the default dialplan: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[2-9]11|0T| 011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT dialplan.digitmap.timeOut=3/ server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /dialplan Jim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec negotiation
We have four settings for the codec. How will it be negotiated? How should it be negotiated in relation to the available bandwidth? Is there an influence by using canreinvite=yes ? Phone A has a setting for the priority of codec Sip.conf has (maybe even different) settings for the priority of codec of this phone A Sip.conf has codec settings for the destination phone B Phone B has (maybe even different) settings for the priority of codec Which codec will be taken? a. if the call goes via * ? b. if the call will be completed with canreinvite=yes ? Which codec should be enforce depending on the bandwidth? Thanks for thinking with me ;-) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 question, take so long time to call
hi all, i use asterisk-1.2.0-beta and asterisk-oh323-0.7.3 with openh323_v1_17_1 . when i dial out with H323, it take so long time to start a 323 call , usually 80 seconds . i havn't used gk. i trace the code , and found out it block at h323_make_call( this function take so long time to return. if i should configure somewhere to let call go more smoothly. i just wanna call like Dial(H323/[EMAIL PROTECTED]) regards, welemon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc and inwards billing
I (cannot sleep and I) am thinking if there is a way to make inwards
billing easy possible.
To dial out we use something like:
exten =
_9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF})
(I have an extra field TARIFF, what allows me to use different prices
for different users)
To dial to a phone we use something like:
exten = 8,1,Dial(SIP/6001,20,tr)
Can we use something like:
exten = 8,4,DeadAGI(astcc.agi,${EXTEN},6001,${TARIFF},3)
and use TARIFF with all location as x cents ?
bye
Ronald Wiplinger
___
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Re: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Mon, 2006-04-17 at 13:19 -0400, Andrew Kohlsmith wrote: Did the number you called (6600546) have a match in the dialplan context it's entering in to (office)? -A. It's supposed to going outbound. [office] is the internal extensions context. And no, it wouldn't find a match there. Jim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardware for new office suggestion
I want to thank you for the suggestions. The office is in the UK, so probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for the line so that bandwidth should not be a problem, the internal LAN will be Gbit as said so the QoS as suggested will be only on the firewall (linux). I have lowered expenses for other equipment so I was thinking of buying a Dell 1800 or 2800 server 2x2,8Ghz 2gb ram to set up Asterisk, know this is a big server but they'll use the ISDN lines and VoIP so virtually there could be 20/25 simultaneous calls. I'll have a look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. Thanks again Simone Tim Panton wrote: On 14 Apr 2006, at 11:29, Simone wrote: Hi list, I am in the process of setting up Asterisk for a new office and since this is going to be my first real installation I'd appreciate some advice on the hardware from the real world. We will have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS on the switch and firewall?), to be configured for both traditional and VoIP usage . I was looking at the Xorcom TS-1 server and I was wondering if you would recommend it for a 30 employees office or if you'd rather build it on a normal server (would a double PIII 1Ghz be enough), and also if you could give a suggestion on the phones (we will get an HP Gbit switch PoE). Thanks, any hint really appreciated Simone I can only base my advice on what we have done for a smaller office. If you want 8 lines it is probably as cheap to go for ISDN 30 as for 4xBRI at least it is here in the UK. We have a single span E1 card from digium without echo can in a small 1U rack mounted server (spec: 1Ghz Via processor and 512Mb ram). The Via might be a bit underpowered for 30 users, but unless you are transcoding, virtually any modern processor would be fine for 8 lines. You need to look out on the DSL line if it is ADSL, since they have low upstream bandwidth. Heavy outgoing mail messages (eg attachments sent to distribution lists) can easily fill the outgoing (256kbit/s) pipe degrading the voice quality. I'm very fond of the SNOM phones - elmeg are selling the old SNOM 190 model which is a decent office phone. For 30 you should be able to get them for less than £70 each. I've got 6 - 4 SNOMs and 2 elmegs - No problems with any of them, but they don't support PoE, so you may want to look at other models. Don't underestimate how much training/doc you will need to provide to get people going on the new system. They may have been using the old one for years and written little cribsheets about how to transfer etc. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Orative
Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php Its seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though Im sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade
On Monday 17 April 2006 14:50, Jim Rice wrote: It's supposed to going outbound. [office] is the internal extensions context. And no, it wouldn't find a match there. How do you have your polycoms set up such that different numbers go to different parts of the dialplan? Every single installation I've ever seen has the phone go to ONE place in the dialplan, and then Asterisk is responsible for figuring out how to get the call to go to where it's supposed to. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardware for new office suggestion
On 4/17/06, Simone [EMAIL PROTECTED] wrote: look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. I just have a few suggestions on the phones.. First of all, try using 1 model for everybody. This makes life much easier in case of upgrades/configuration/central provisioning, etc.. Some phones I have used (with success) are: Polycom 501, Thomson ST2030, Cisco 7940/7660. cheers and good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Orative
Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message-From: Dean Collins [mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php Its seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though Im sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Orative
The weather isn't as good in Indiana. Douglas Garstang wrote: Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- *From:* Dean Collins [mailto:[EMAIL PROTECTED] *Sent:* Monday, April 17, 2006 1:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It’s seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I’m sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CentOS 4.x Asterisk RPMS
Erick, That's it for now. Let me know offlist if you have any requests... Andrew Erick Perez wrote: Andrew, the only two patches are the ones you mention here?(spandsp and iaxmodem)? no other patches? Thanks, On 4/15/06, alist [EMAIL PROTECTED] wrote: I have complied the latest releases, patched with spandsp and iaxmodem ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Orative
And they have better sushi than Montana and Wyoming -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew D Kirch Sent: Monday, April 17, 2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Orative The weather isn't as good in Indiana. Douglas Garstang wrote: Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- *From:* Dean Collins [mailto:[EMAIL PROTECTED] *Sent:* Monday, April 17, 2006 1:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It's seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I'm sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail use external smtp server for sendingmail
I had the same question, and I want to make sure I'm clear. This implies to me that Asterisk itself doesn't use SMTP, but rather dumps a message into some directory that Sendmail on the same box will see and process? I have no problem getting Sendmail to use a smarthost, but am I understanding the Asterisk part of this properly, or is there a way to get Asterisk to DIRECTLY use a smarthost, so that Sendmail doesn't have to be running on the local Asterisk box? Thanks! -Steve -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Saturday, April 15, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail use external smtp server for sendingmail Yes, just configure your sendmail to do it. On 4/13/06, nik600 [EMAIL PROTECTED] wrote: is it possibile to set up an external smtp server for the relay to the users of the mails? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astcc and inwards billing
Ronald Wiplinger wrote:
I (cannot sleep and I) am thinking if there is a way to make inwards
billing easy possible.
To dial out we use something like:
exten =
_9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF})
(I have an extra field TARIFF, what allows me to use different prices
for different users)
To dial to a phone we use something like:
exten = 8,1,Dial(SIP/6001,20,tr)
Can we use something like:
exten = 8,4,DeadAGI(astcc.agi,${EXTEN},6001,${TARIFF},3)
and use TARIFF with all location as x cents ?
Short answer is Yes.
I use a similar astcc.agi call for in-network calls.
The cost will depend on how you've implemented TARIFF and whether it
adds to or bypasses the cost field for your 600x SIP route.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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Re: [Asterisk-Users] Orative
Andrew D Kirch wrote: The weather isn't as good in Indiana. Hear hear!! There had to have been a hundred thousand lightning strikes last night within a mile of my house. . . B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Orative
Easy access to talent.. and lets face it where else are you going to start up? NY.. I hate it and I have to live here. BTW another neat start up I saw at CTIA last week is www.savaje.com Java based mobile OS being sold to MVNO on other companies hardware, highly customisable and OTA updates, very neat one to watch. Based in MA.does that make you happier J Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Monday, 17 April 2006 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Orative Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Monday, April 17, 2006 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php Its seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though Im sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI Installation Eicon Diva Server
Armin Schindler wrote: The configuration is as easy as with BRI lines. Can you provide more (like your confs and verbose/debug output)? Also (this isn't directed at you Armin, but I found your email to reply off of to maintain the threading), I created a Wiki page over at the freePBX documentation site, explaining how to configure an Eicon Server 4-BRI for freePBX. It may have some tips for you: http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva Feel free to add/remove information. Its a Wiki after all. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Orative
Southern California would make me happy, maybe the north west. :) -Original Message-From: Dean Collins [mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:49 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Orative Easy access to talent.. and lets face it where else are you going to start up? NY.. I hate it and I have to live here. BTW another neat start up I saw at CTIA last week is www.savaje.com Java based mobile OS being sold to MVNO on other companies hardware, highly customisable and OTA updates, very neat one to watch. Based in MA.does that make you happier J Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Monday, 17 April 2006 3:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Orative Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message-From: Dean Collins [mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php Its seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though Im sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hyperthreading compiling.
Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Orative
I resent that, the weather here is wonderful today On 4/17/06, Andrew D Kirch [EMAIL PROTECTED] wrote: The weather isn't as good in Indiana. Douglas Garstang wrote: Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- *From:* Dean Collins [mailto:[EMAIL PROTECTED] *Sent:* Monday, April 17, 2006 1:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It's seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I'm sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hyperthreading compiling.
Wai Wu wrote: Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx Wai, Asterisk makes heavy use of threads, so all that's required to take advantage of HyperThreading is an SMP kernel. The kernel itself will take care of scheduling the threads on the different virtual and (if they exist) physical CPUs. In my experience, Asterisk scales well across processors. On my production server (a Quad Xeon with HyperThreading turned off), 'ps auxm' currently shows 160 Asterisk threads, none of which are taking over 0.8% of CPU. Keep in mind that HyperThreading does not guarantee better performance. Depending on the application, you may actually see a performance degradation http://news.zdnet.co.uk/0,39020330,39237341,00.htm. I've heard arguments on both sides of the HyperThreading debate as it pertains to Asterisk, so my best advice is to do some testing and see which works best for your particular scenario. Remember, what may be good for CPU utilization might be bad for the quality of a call using Digium hardware. Input from other users may also be very valuable in helping you decide whether or not to enable HyperThreading. Providing a more detailed description of your server, including the Digium hardware installed, would probably lead to more constructive information. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hyperthreading compiling.
I highly recommed against using hyperthreading. It always seems to cause intermittent kernel panics for me when I forget to turn it off. -- Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hyperthreading compiling.
We've been running Asterisk on P4s with HyperThreading turned on with an SMP kernel for almost two years now. Currently 12 production servers, No problems and slightly higher capacity. MATT--- On 4/17/06, Justin Tunney [EMAIL PROTECTED] wrote: I highly recommed against using hyperthreading. It always seems to cause intermittent kernel panics for me when I forget to turn it off. -- Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk settings for roaming users
Hi, like to know which configurations are most suitable for roaming users accessing from various external environments? As an example, should I use nat=yes in sip.conf when the end device could be connecting from behind nat with private ip or with a public ip? Appreciate any suggestions. Thanks. Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - Accessible with your email software or over the web ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
