[Asterisk-Users] false hangups
Hello,
We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume
environment. At least twice a day there are complaints of 'dropped calls'.
Examining the debug logs, I see that in each case, an on hook event is
detected, followed by the zap channel being hung-up and * saying BYE to the
sip phone:
Jun 23 14:17:22 DEBUG[2441232]: Exception on 22, channel 1
Jun 23 14:17:22 DEBUG[2441232]: Got event On hook(1) on channel 1 (index 0)
We are using fxs_ks, and neither 'callprogress' nor 'busydetect 'are being
used.
What could possibly signal these hangups?
On each of the incoming analog lines, there are splitters to unused analog
phones ('bat-phones' .. just in case) ... is it possible that this is the
source of the problem?
Zaptel from 2004.06.14 CVS.
Asterisk from 2004.06.02 CVS
Thank you
--
..
Ryan Courtnage
Coalescent Systems
403.830.9410
[EMAIL PROTECTED]
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re: [Asterisk-Users] Which Linux ?
From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. Tim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Which Linux ?
Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel 2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked perfectly both times, without needing any additional compiler flags, and no kernel panics. What I have found out is that I had to disable hyperthreading, or I would be getting very choppy audio (I think that's what you mean when you say needs kernel thread turned off). By the way, the noht flag in lilo/grub isn't enough, it has to be disabled in the BIOS. Don't know if that's an issue on other Linuxes as well. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Read this if you use CVS HEAD (Unstable) and chan_capi and jitter buffering in IAX
Hi, IF: you are using chan_capi in combination with IAX, and you are using the IAX jitter buffer (see iax.conf) and you run with the CVS HEAD version of Asterisk, THEN: You must make sure that you are running the most current chan_capi version - which is 0.3.4 at the time of writing. Older versions of chan_capi don't initialise an important timestamp in audio frames - with the result that capi originated calls forwarded over IAX will probably end up with no audio. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?
hi Jakob¸ i see you have installed 2 fritz card in PC .. I have a lot of problems already with one card .. When i type capiinfo computer freeze. I have shared irq of fritz card with some motherboard resources ? You think this is a problem ? what kernel you use? what drivers version of drivers you have? Could you send me patched drivers for second card (sources) - i already patch two times but no success :( you have now working asterisk with two cards? thank you, Tomaz Jakob Strebel wrote: Hi, I tried to install the following hack. http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO But the 2nd AVM Fritz PCI card is still not showing up. My environment is: debian 2.4.24 (asterisk 0.72) Just a quick explanation what I did: - edited the filed as described above - make clean - make - make install - reboot my machine - modprobe capi - insmod -f fcpci What do I miss? What did I do wrong? regards jakob asterisk:~# lsmod Module Size Used byTainted: PF fcpci 532320 2 capi6528 4 kernelcapi 30624 3 [fcpci capi] capiutil 22912 0 [kernelcapi] asterisk:~# asterisk:/var/log# capiinit start modprobe: Can't locate module capifs modprobe: Can't locate module capifs WARNING: filesystem capifs not available modprobe: Can't locate module f2pci ERROR: failed to load driver f2pci asterisk:/var/log# This is what I see in the log file: Mar 22 17:37:55 asterisk -- MARK -- Mar 22 17:42:14 asterisk kernel: CAPI-driver Rev 1.1.4.1: loaded Mar 22 17:42:14 asterisk kernel: capi20: started up with major 68 Mar 22 17:42:14 asterisk kernel: kcapi: capi20 attached Mar 22 17:42:14 asterisk kernel: capi20: Rev 1.1.4.2: started up with major 68 (no middleware) Mar 22 17:42:41 asterisk kernel: fcpci: AVM FRITZ!Card PCI (2nd) driver, revision 0.5.2 Mar 22 17:42:41 asterisk kernel: fcpci: (fcpci built on Mar 22 2004 at 17:41:36) Mar 22 17:42:41 asterisk kernel: fcpci: Loading... Mar 22 17:42:41 asterisk kernel: f2pci: Driver 'fcpci' attached to stack Mar 22 17:42:41 asterisk kernel: kcapi: driver fcpci attached Mar 22 17:42:41 asterisk kernel: fcpci: Auto-attaching... Mar 22 17:42:41 asterisk kernel: PCI: Enabling device 00:0f.0 ( - 0003) Mar 22 17:42:41 asterisk kernel: PCI: Assigned IRQ 5 for device 00:0f.0 Mar 22 17:42:41 asterisk kernel: fcpci: Stack version 3.11-02 Mar 22 17:42:41 asterisk kernel: kcapi: Controller 1: fritz2-pci attached Mar 22 17:42:41 asterisk kernel: kcapi: card 1 fritz2-pci ready. Mar 22 17:42:41 asterisk kernel: fcpci: Loaded. Mar 22 17:42:41 asterisk kernel: kcapi: notify up contr 1 Mar 22 17:42:41 asterisk kernel: capi: controller 1 up Mar 22 17:57:55 asterisk -- MARK -- asterisk:/var/log# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Which Linux ?
On 24/06/2004, at 4:48 PM, Manuel Wenger wrote: Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel 2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked perfectly both times, without needing any additional compiler flags, and no kernel panics. I'm running two production systems on Fedora Core 1 with the 2.4.22-1.2188 kernel - one with Digium hardware and one running zaprtc CAPI cards. Both systems run fine, compiled straight out of the box, and as of yet I have had no issues. I've also run it quite happily on RedHat 8 under VirtualPC with no additional hardware. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video/H323/SIP
Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice ip10s
Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDRs, Conferencing, and MeetMe
Hi, -Original Message- As teleconferencing is the only application of the Asterisk box, I have the dialplan setup to immediately launch into the MeetMe application and prompt the user for conference number/PIN upon answering. It appears that the MeetMe module isn't interested in passing the conference number back to Asterisk when the user disconnects so that Asterisk can include that information in the CDR. Any suggestions on how to do this? How about not dropping them straight into the Meetme, but give Meetme parameters based on what the user enters in an Authenticate. If you add the 'a' parameter the password (conference room) is stored in the accountcode (which is in CDR). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swissvoice ip10s
Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Jun 24, 2004, at 12:33 AM, Florian Overkamp wrote: Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary> And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Read this if you use CVS HEAD (Unstable) and chan_capi and jitter buffering in IAX
version - which is 0.3.4 at the time of writing. 0.3.4a since more than a week. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway
provider supports both, so that's not a problem, and if I force him (the PSTN gateway)
to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a
first priority, then G729. I did it like that:
[mypstngate]
type=friend
host=192.168.0.100
port=5060
context=pstn-in
canreinvite=no
disallow=all
allow=ulaw
allow=g729
Then, in the outgoing context for our G729 SIP customers, I've put something like that:
exten = _0N,1,setvar(SIP_CODEC=g729)
exten = _0N,2,Dial(SIP/0041${EXTEN:[EMAIL PROTECTED],90)
What happens now when placing a call is very interesting. As you can see, Asterisk
wants to change the codec to g729, but on the outgoing call to the PSTN gateway it
remains ULAW. Like this, I'm using up one of my G729 licenses, and Asterisk is doing
the transcoding between G729 and ULAW. That's definitely not what I want. Any ideas
about how to force both channels to G729? By the way, if I use a client which doesn't
support G729, this call doesn't even take place, it hangs up, because Asterisk tries
to force G729 on the client's channel (but not on the PSTN gateway's channel).
In other words, the setvar(SIP_CODEC=g729) only forces the codec on the calling
channel, not on the called channel. How can I change that?
Another interesting thing, the show g729 after the call hangs up: I have -1/-2
encoders/decoders in use. Maybe a bug?
Thanks
-Manuel
*CLI -- Executing SetVar(SIP/2016-b119, SIP_CODEC=g729) in new stack
-- Executing Dial(SIP/2016-b119, SIP/[EMAIL PROTECTED]|90) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/mypstngate-caed is making progress passing it to SIP/2016-b119
-- SIP/mypstngate-caed is ringing
-- SIP/mypstngate-caed answered SIP/2016-b119
Jun 24 09:49:23 NOTICE[1094450096]: chan_sip.c:1314 sip_answer: Changing codec to
'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed
*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
192.168.0.1000041911234 1f7d34e3642 00102/0 0ms ms ULAW
192.168.0.2 20164977-4F41-7 00101/3 0ms ms G729A
2 active SIP channel(s)
[... after hangup ...]
== Spawn extension (auth-out, 0911234567, 2) exited non-zero on 'SIP/2016-b119'
-- Executing Hangup(SIP/2016-b119, ) in new stack
== Spawn extension (auth-out, h, 1) exited non-zero on 'SIP/2016-b119'
cdr_odbc: Query Successful!
*CLI show g729
-1/-2 encoders/decoders of 30 licensed channels are currently in use
*CLI
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[Asterisk-Users] 2 E100P cards on one asterisk
Hello, Is it possible to have 2 E100P cards on one asterisk? I'm able to do that now, but I'm not sure about my config. Here is my zaptel.conf [EMAIL PROTECTED] asterisk]# more ../zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr span=1,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone=fr defaultzone=fr Here is my zapata.conf [EMAIL PROTECTED] asterisk]# more zapata.conf [channels] switchtype=national context=default signalling=pri_net group=1 channel = 1-15,17-31 switchtype=national context=default signalling=pri_cpe group=2 channel = 32-46,48-62 But I get such errors in the console: -- Called g1/600 Jun 24 03:16:34 WARNING[1167272000]: chan_zap.c:6655 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet -- Executing Playback(Zap/32-1, demo-echotest) in new stack Thanks. Fred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
We use Red Hat Enterprise Linux 3 and Debian 3.0 r2 (woody) for our commercial installations. Asterisk runs well on both but you will find RHEL3 much easier to install and find that it supports a wider range of hardware configurations without requiring the kernel builds that a Debian install frequently involves. Indeed, with some more unusual hardware (i.e. IBM ServeRaid controllers) one has to build a new Debian kernel on a separate machine to support the controller whereas RHEL3 installs straight of the CDs. On the other hand, the obvious cost benefit of Debian can compensate for the steeper learning curve when one is doing dozens of installs. The problem with Red Hat 8 or 9 is that support and fixes have been discontinued. For hobbyist installations that may not be an issue but for our customers they are non-starters. This website provides guidance on installing Asterisk on Debian: http://users.pandora.be/Asterisk-PBX/index.htm George Pajari netVOICE communications www.netvoice.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay in Zap Calls?
I have this line in my extensions.conf,
exten = _393.,1,Dial(ZAP/${EXTEN:3},20,tr)
when I make a zap call, it gives me two rings and then makes the zap call.
Is there is a way I can make the call immediate?
Kannaiyan
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[Asterisk-Users] Swissv oice IP10 behind NAT
Hi, Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already use this fonctionnality with a PBX other then asterisk, and it work very well. In fact, I need to use this settings in the IP10 telnet session set xgcp rgw_name MAC_ADDR set tcid 0 notify_entity [EMAIL PROTECTED]@IP PBX]:2427 In fact, the phone register with the MAC@ and not with the IP@, and the PBX reply the phone in the UDP or TCP open channel on the router or on the firewall. A channel is always open by the phone, so there is no problem. Is this fonctionnality possible with asterisk? If not yet, is it planned? Thanks. Fred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot register Zyxel Prestige 2000W
Hello, I am having trouble registering a SIP telephone Zyxel Prestige 2000W to my Asterisk PBX. I am rather confident that Asterisk is working as there are other H323 telephones working fine and a couple of SJPhones working correctly. I keep getting: Jun 24 10:31:08 NOTICE[-1254138960]: chan_sip.c:6742 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.10.3.82' while my sip.conf has a section like this [898] type=friend username=phone secret=rombo callerid=Portatile SIP 898 host=dynamic canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw context=sip I even tried without the username=, but I get the same message. Anybody can help me? Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to force G729
allow=ulaw Why don't you remove this? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot register Zyxel Prestige 2000W
Hi I got it to work straight off (i've since upgraded the code to WJ.00.0b-t04) sip.conf [1003] type= friend username=1003 secret=blah host=dynamic context=from-sip dtmfmode=rfc2833 Zyxel config SIP/outbound Proxy config Proxy IP:192.168.254.1 Proxy port = 5060 SIP Config SIP UII sip: 1003 @ 192.168.254.1: 5060 Expire time 300 Registrar username 1003 display name 1003 DSP setting Default Voice codec G.729 8k DTMF relay outbound Hope this helps Cheers Giles - Original Message - From: lenz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 10:10 AM Subject: [Asterisk-Users] Cannot register Zyxel Prestige 2000W Hello, I am having trouble registering a SIP telephone Zyxel Prestige 2000W to my Asterisk PBX. I am rather confident that Asterisk is working as there are other H323 telephones working fine and a couple of SJPhones working correctly. I keep getting: Jun 24 10:31:08 NOTICE[-1254138960]: chan_sip.c:6742 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.10.3.82' while my sip.conf has a section like this [898] type=friend username=phone secret=rombo callerid=Portatile SIP 898 host=dynamic canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw context=sip I even tried without the username=, but I get the same message. Anybody can help me? Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: -- [Asterisk-Users] Serious issues with current CVS?
I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at all (ie the stuff below:) The previous cvs download from a few weeks ago (June 9th) compiled and loaded fine. /* We now use the linux kernel config to detect which options to use */ /* You can still override them below */ /* #if defined(CONFIG_HDLC) || defined(CONFIG_HDLC_MODULE) #define CONFIG_ZAPATA_NET #if LINUX_VERSION_CODE = KERNEL_VERSION(2,4,20) #define CONFIG_OLD_HDLC_API #else #if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,3) #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT #endif #endif #endif #ifdef CONFIG_PPP #define CONFIG_ZAPATA_PPP #endif */ They also seem to have broken the SayUnixTime app - it can't cope with a 'digits/at' bit in the middle of the string any more. I have had to change exten = 123,1,SayUnixTime(||AdBY 'digits/at' IM) to exten = 123,1,SayUnixTime(||AdBY) exten = 123,2,Playback(digits/at) exten = 123,3,SayUnixTime(||IM) Peter -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: 24 June 2004 06:45 To: Asterisk List Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? For me chan_capi set up the call but no sound available. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Try to configure in sip.conf your extensions context like this: [XXX] disallow=all allow=g729 Done that already: but then, the incoming channel (from the user to Asterisk) is G729, and the outgoing channel (from Asterisk to the PSTN gateway) still remains ULAW, so Asterisk has to do transcoding (ULAW to G729), which I don't want, obviously. For some reason, the outgoing channel doesn't follow the setvar(SIP_CODEC=g729) rule. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: [Asterisk-Users] How to force G729
Define that per user. Of course... The user part is not the problem. If I force a user in its extensions to use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the PSTN gateway, doing the transcoding. This is driving me crazy... -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot register Zyxel Prestige 2000W
Hello Giles, thank you, in fact it did! Not sure why, had to use a number for the username and play with the codec. Now I can receive calls on the Zyxel (but sometimes crashes with a 500 Interbnal server error, as Asterisk reports) and I cannot place calls from the Zyxel (not sure why, makes the same tritone as the mobile when it's impossible to place a call). Would you mind if I put your recipe on voip-info? Thanks, l. In data Thu, 24 Jun 2004 10:34:48 +0100, Giles Scott [EMAIL PROTECTED] ha scritto: Hi I got it to work straight off (i've since upgraded the code to WJ.00.0b-t04) sip.conf [1003] type= friend username=1003 secret=blah host=dynamic context=from-sip dtmfmode=rfc2833 Zyxel config SIP/outbound Proxy config Proxy IP:192.168.254.1 Proxy port = 5060 SIP Config SIP UII sip: 1003 @ 192.168.254.1: 5060 Expire time 300 Registrar username 1003 display name 1003 DSP setting Default Voice codec G.729 8k DTMF relay outbound Hope this helps Cheers Giles - Original Message - From: lenz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 10:10 AM Subject: [Asterisk-Users] Cannot register Zyxel Prestige 2000W Hello, I am having trouble registering a SIP telephone Zyxel Prestige 2000W to my Asterisk PBX. I am rather confident that Asterisk is working as there are other H323 telephones working fine and a couple of SJPhones working correctly. I keep getting: Jun 24 10:31:08 NOTICE[-1254138960]: chan_sip.c:6742 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.10.3.82' while my sip.conf has a section like this [898] type=friend username=phone secret=rombo callerid=Portatile SIP 898 host=dynamic canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw context=sip I even tried without the username=, but I get the same message. Anybody can help me? Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with chan_capi
Hi all, I´m a newbie @ asterisk and i´m getting in trouble while configuring asterisk for ISDN first run. I´m using Debian testing an the standard packages of asterisk and chan_capi. My Fritz Card Capis are working fine. I´m already using i4l and hylafax on this machine. When I include the chan_capi to the modules.conf I´m getting an Error like this: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Illegal instruction (core dumped) Asterisk ist no stopped. My modules.conf looks like: [modules] autoload=no noload => pbx_gtkconsole.so noload => pbx_kdeconsole.so noload => app_intercom.so noload => chan_modem.so noload => chan_modem_i4l.so noload => chan_modem_bestdata.so noload => chan_modem_aopen.so load => res_musiconhold.so load => res_parking.so load => chan_capi.so noload => chan_iax2.so noload => chan_zap.so noload => chan_alsa.so noload => chan_oss.so [global] chan_capi.so=yes And the capi.conf is: [general] nationalprefix=0 internationalprefix=00 [interfaces] msn=9142829 incomingmsn=9142829 controller=1 softdtmf=1 devices=2 Does anyone have an idea whats going wrong here? THX4help Markus PGP.sig Description: Signierter Teil der Nachricht
Re: R: [Asterisk-Users] How to force G729
When I set the SIP_CODEC variable to force g729:
Jun 24 12:30:01 NOTICE[1226062640]: chan_sip.c:1313 sip_answer: Changing
codec to 'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/8011-86fe and SIP/8008-c2b9
Jun 24 12:30:02 NOTICE[1234455344]: channel.c:1508 ast_set_read_format:
Unable to find a path from G729A to ULAW
Jun 24 12:30:02 NOTICE[1234455344]: channel.c:1478 ast_set_write_format:
Unable to find a path from ULAW to G729A
Jun 24 12:30:02 WARNING[1226062640]: chan_sip.c:1332 sip_write: Asked to
transmit frame type 256, while native formats is 4 (read/write = 4/4)
== Spawn extension (sip, 8041, 2) exited non-zero on 'SIP/8011-86fe'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back
from 217.117.xxx.xxx
Though I get a short 'hello' (voice) from the otherside, but after that
line dies.
Stefan
On Thu, 24 Jun 2004, Manuel Wenger wrote:
Try to configure in sip.conf your extensions context like this:
[XXX]
disallow=all
allow=g729
Done that already: but then, the incoming channel (from the user to Asterisk) is
G729, and the outgoing channel (from Asterisk to the PSTN gateway) still remains
ULAW, so Asterisk has to do transcoding (ULAW to G729), which I don't want,
obviously.
For some reason, the outgoing channel doesn't follow the setvar(SIP_CODEC=g729) rule.
-Manuel
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Re: [Asterisk-Users] Cannot register Zyxel Prestige 2000W
Hi Glad I could help. sure please feel free to post. Cheers Giles - Original Message - From: lenz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 11:29 AM Subject: Re: [Asterisk-Users] Cannot register Zyxel Prestige 2000W Hello Giles, thank you, in fact it did! Not sure why, had to use a number for the username and play with the codec. Now I can receive calls on the Zyxel (but sometimes crashes with a 500 Interbnal server error, as Asterisk reports) and I cannot place calls from the Zyxel (not sure why, makes the same tritone as the mobile when it's impossible to place a call). Would you mind if I put your recipe on voip-info? Thanks, l. In data Thu, 24 Jun 2004 10:34:48 +0100, Giles Scott [EMAIL PROTECTED] ha scritto: Hi I got it to work straight off (i've since upgraded the code to WJ.00.0b-t04) sip.conf [1003] type= friend username=1003 secret=blah host=dynamic context=from-sip dtmfmode=rfc2833 Zyxel config SIP/outbound Proxy config Proxy IP:192.168.254.1 Proxy port = 5060 SIP Config SIP UII sip: 1003 @ 192.168.254.1: 5060 Expire time 300 Registrar username 1003 display name 1003 DSP setting Default Voice codec G.729 8k DTMF relay outbound Hope this helps Cheers Giles - Original Message - From: lenz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 10:10 AM Subject: [Asterisk-Users] Cannot register Zyxel Prestige 2000W Hello, I am having trouble registering a SIP telephone Zyxel Prestige 2000W to my Asterisk PBX. I am rather confident that Asterisk is working as there are other H323 telephones working fine and a couple of SJPhones working correctly. I keep getting: Jun 24 10:31:08 NOTICE[-1254138960]: chan_sip.c:6742 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.10.3.82' while my sip.conf has a section like this [898] type=friend username=phone secret=rombo callerid=Portatile SIP 898 host=dynamic canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw context=sip I even tried without the username=, but I get the same message. Anybody can help me? Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with chan_capi
Title: Mensaje send us a debug file, but first Have you load CAPI driver for Fritz? Avanzada 7, S.L. Sergio Serrano RevueltoRD Manager Avda. Juan López de Peñalver 17Edificio Centro de Empresas Planta 3ª, Pasillo BParque Tecnológico de Andalucía29590 Campanillas(Málaga) [EMAIL PROTECTED] tel: tel2:fax: mobile: (+0034) 951014947(+0034) 951014943. Ext 705(+0034) 951010922618747717 Signature powered by Plaxo Want a signature like this? Add me to your address book... -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Markus KleinEnviado el: jueves, 24 de junio de 2004 12:37Para: [EMAIL PROTECTED]@[EMAIL PROTECTED]Asunto: [Asterisk-Users] Help with chan_capiHi all,I´m a newbie @ asterisk and i´m getting in trouble while configuring asterisk for ISDN first run. I´m using Debian testing an the standard packages of asterisk and chan_capi. My Fritz Card Capis are working fine. I´m already using i4l and hylafax on this machine. When I include the chan_capi to the modules.conf I´m getting an Error like this:[chan_capi.so] = (Common ISDN API for Asterisk)== Parsing '/etc/asterisk/capi.conf': FoundIllegal instruction (core dumped)Asterisk ist no stopped. My modules.conf looks like:[modules]autoload=nonoload = pbx_gtkconsole.sonoload = pbx_kdeconsole.sonoload = app_intercom.sonoload = chan_modem.sonoload = chan_modem_i4l.sonoload = chan_modem_bestdata.sonoload = chan_modem_aopen.soload = res_musiconhold.soload = res_parking.soload = chan_capi.sonoload = chan_iax2.sonoload = chan_zap.sonoload = chan_alsa.sonoload = chan_oss.so[global]chan_capi.so=yesAnd the capi.conf is:[general]nationalprefix=0internationalprefix=00[interfaces]msn=9142829incomingmsn=9142829controller=1softdtmf=1devices=2Does anyone have an idea whats going wrong here?THX4helpMarkus
[Asterisk-Users] SIP clients, H323 client as gateway?
I have several SIP clients which connect to Asterisk. Is it possible to use a H323 client account (username/password combination) as a gateway/proxy for these SIP clients when they need to call someone on this H323 network, just like using an IAX2-gateway (registering an username/password combi). I took a look in the archives, but couldn't find anything related. But then again, I might not have looked carefully enough ;) Thanks, Ralf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL Prestige 2000W and DTMF
I've just seen this post: http://www.mail-archive.com/[EMAIL PROTECTED]/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF
Hi, With my config (as posted this morning) DTMF works. I can log onto voicemail by selecting a mailbox number and password Giles - Original Message - From: Dominique Kull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:02 PM Subject: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF I've just seen this post: http://www.mail-archive.com/[EMAIL PROTECTED]/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
The echo fix went into Head cvs late yesterday (CDT). I don't have an x100p installed right now, but I'd strongly suggest doing a new checkout and test it. Greater than 70% chance its fixed. Beside of problem of CallerID not working in several countries, what specially I still don't know if it's a X100P's limitation or Zap driver limitation, the major problem with this devices is no doubt the echo. If you guys can solve this with tdms and x100ps, it's gonna be a big step. On Wed, 23 Jun 2004, Rich Adamson wrote: Good news! FYI, worked with Mark this afternoon to test changes needed to reduce or eliminate echo involving pstn calls on the new tdm fxo card (bug #1902). Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the changes were only to chan_zap.c, I'll have to guess it will improve the historic x100p echo issues as well, and probably other zap channels also. The changes were associated with the echo cancel routines, but as a non-programmer, I'm not sure what these changes really do (other than the end result). If these changes prove to be successfull, then Mark is going to make them configurable options somewhere. Got'a love it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
On Jun 24, 2004, at 2:33 AM, [EMAIL PROTECTED] wrote: From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. I asked a similar question about SuSE 9.1 and the 2.6 kernel a little while ago: http://lists.digium.com/pipermail/asterisk-users/2004-May/048781.html Follow the thread out and it seems some have had success. I have been doing our preliminary testbed with SuSE 9.0 and it is working just fine. Our final deployment will be under 9.1 and the 2.6 kernel. I already have * and zaptel built on that system, but we haven't run it yet. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP hangup not working with siemens HICOM
Hello everybody, any configure asterisk with siemens HICOM, or HIPATH. I am facing asevere hangup issue. two servers at station-1 and station-2 both with 8 fxo lines. extensions from siemens are plugged into the fxo cards. now problem is that, if call is made from station-1 to station-2 and then if caller hanged up, 80% chances are that asterisk will not hangup the call, it will remain stucked in the asterisk.\ will playing with the busy tone in indications.conf sort out this problem? Thanks Atif Sent via the WebMail system at convergence.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anonymity and Privacy headers
When a user calling over the PSTN network calls one of our SIP users with a restricted number (CLIR), our PSTN gateway is sending us incoming calls with the following additional headers: Proxy-Require: privacy Anonymity: uri Remote-Party-ID: sip:[EMAIL PROTECTED]:4000;privacy=uri as opposed to Remote-Party-ID: sip:[EMAIL PROTECTED]:4000;privacy=off when CLIP is enabled (thus CLIR is disabled). Any ideas on how I can tell asterisk to process one (or more) of these headers, and strip the CLI before sending the call out to our SIP users, in case it is restricted? I have searched the Wiki and read the chan_sip.c source code, but didn't find anything useful... Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with chan_capi
Illegal instruction (core dumped) This has nothing to do with ISDN or CAPI or whatever. Either a) your board or memory is buggy b) you have a 486 and compiled for 586 or 686 c) you have a 686 and compiled for 686 etc ... b) or c) usually happens when you use Asterisk on some very small (almost embedded) main board, not with a Standard ATX board with a Standard Athlon or Pentium CPU. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
Mike, I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? TIA Julian - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:30 PM Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 2:33 AM, [EMAIL PROTECTED] wrote: From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. I asked a similar question about SuSE 9.1 and the 2.6 kernel a little while ago: http://lists.digium.com/pipermail/asterisk-users/2004-May/048781.html Follow the thread out and it seems some have had success. I have been doing our preliminary testbed with SuSE 9.0 and it is working just fine. Our final deployment will be under 9.1 and the 2.6 kernel. I already have * and zaptel built on that system, but we haven't run it yet. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF
I have tested the exact same config, but had no luck. I managed to get it going with some different settings on the phone, though. ZyXEL settings: DTMF RELAY inband(RFC2833) ?? DTMF Payload 101 ?? for the sip.conf (same as Giles apart from forcing g.729) [400] type=friend username=400 secret=blah host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g729 callerid=Vintage Cell Phone 400 It is all a bit confusing regarding what is inband and outband on the phone. I am also not sure about DTMF Payload type... but it seems to work ok. regards Dominique Giles Scott wrote: Hi, With my config (as posted this morning) DTMF works. I can log onto voicemail by selecting a mailbox number and password Giles - Original Message - From: Dominique Kull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:02 PM Subject: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF I've just seen this post: http://www.mail-archive.com/[EMAIL PROTECTED]/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video/H323/SIP
Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
On Jun 24, 2004, at 7:55 AM, Asterisk wrote: I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? Well, I will tell you, but I'll then have to kill you... Lucky for you I have no clue who you are or where you reside :) The incantation I used is: make linux26 That machine isn't running right now so I can't double-check it, but I don't recall having to do anything out of the ordinary. I do remember banging my head for a while before I remembered that there is a different build sequence for the 2.6 kernel... If this doesn't work for you, let me know and when I have a chance I'll fire that system up and give it another go-'round... - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:30 PM Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 2:33 AM, [EMAIL PROTECTED] wrote: From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. I asked a similar question about SuSE 9.1 and the 2.6 kernel a little while ago: http://lists.digium.com/pipermail/asterisk-users/2004-May/048781.html Follow the thread out and it seems some have had success. I have been doing our preliminary testbed with SuSE 9.0 and it is working just fine. Our final deployment will be under 9.1 and the 2.6 kernel. I already have * and zaptel built on that system, but we haven't run it yet. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video/H323/SIP
I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: [Asterisk-Users] How to force G729
Define that per user. Of course... The user part is not the problem. If I force a user in its extensions to use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the PSTN gateway, doing the transcoding. This is driving me crazy... If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse
On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Just to further update this issue: Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the changes were only to chan_zap.c, I'll have to guess it will improve the historic x100p echo issues as well, and probably other zap channels also. The changes were associated with the echo cancel routines, but as a non-programmer, I'm not sure what these changes really do (other than the end result). If these changes prove to be successfull, then Mark is going to make them configurable options somewhere. Got'a love it. I've just done the upgrade on my test asterisk system. On a series of outgoing PSTN calls via our TDM31B there was no detectable echo. We no longer need to use aggressive echo cancellation in the zaptel driver either - which is much nicer for our staff when they are making calls. Thanks a bunch for fixing this. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using 2 single pri cards on 1 server
I just set this same configuration up yesterday: 2T100P cards in 1 asterisk box, and yes, it does work.. /etc/zaptel.conf: span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23,25-47 dchan=24,48 On Tue, 22 Jun 2004, jan wrote: I dispose of a asterisk server with a quad pri card in it and a asterisk server with a single pri card. Could I add a second single pri card to the second server ? It is for multiplexing purposes. Regards, Jan -Mike == Network Engineer Pathway Internet Services 616.774.3131 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dead air on 7960 sip at start of call.
Hi, Despite much searching, I can't find anything quite like the problem we seem to be experiencing with our recently activated asterisk pbx. Of four people answering several calls per five minutes, I have reports of the occasional call starting with dead air. The user picks up the call from the queue, and they can hear that the line is too quiet. If they start talking, the caller either doesn't hear them, or only hears the end of the first sentence. Reports are that this silence at the start of the call lasts from 4-5 seconds, sometimes over 10 seconds and more rarely over 20 seconds. After the period of silence the call proceeds with no further problems. I have taken packet dumps of extended periods, there is nothing consistently different about the calls with silent start compared to those without the problem. The SIP INVITE is sent to the phone, the phone rings, and OK's the call. The RTP stream begins straight away. I'm using cvs-head as of 2004-06-10, CVS-D2004.06.09.14.00.00, recently updated from v1-0_stable dated a few weeks before. This problem existed with both versions. The phones are Cisco 7960's with SIP image 6.3, the calls are incoming on an E1 connected to a TE410P. Calls are queued and the phones statically defined as agents in the queue which has a ringall strategy. There is no queue announcement played to the answering user. Asterisk in vvv verbose doesn't log anything different when this occurs. So, has anyone else heard of something like this? Or have a suggestion for further investigation? Cheers. -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: [Asterisk-Users] How to force G729
If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to no, so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using disallow=all; allow=g729 in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I force it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says OK, now Dial(... @gateway), but force G729 (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dead air on 7960 sip at start of call.
Hi, I've noticed the same thing with my 7960 running SIP 7.1. I came to the conclusion that the Cisco does some clever echo cancellation, and that pause is the echo canceller training. I don't think it's an Asterisk problem. My Grandstream 101 works perfectly. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Thu, 24 Jun 2004, Sam Tilders wrote: Hi, Despite much searching, I can't find anything quite like the problem we seem to be experiencing with our recently activated asterisk pbx. Of four people answering several calls per five minutes, I have reports of the occasional call starting with dead air. The user picks up the call from the queue, and they can hear that the line is too quiet. If they start talking, the caller either doesn't hear them, or only hears the end of the first sentence. Reports are that this silence at the start of the call lasts from 4-5 seconds, sometimes over 10 seconds and more rarely over 20 seconds. After the period of silence the call proceeds with no further problems. I have taken packet dumps of extended periods, there is nothing consistently different about the calls with silent start compared to those without the problem. The SIP INVITE is sent to the phone, the phone rings, and OK's the call. The RTP stream begins straight away. I'm using cvs-head as of 2004-06-10, CVS-D2004.06.09.14.00.00, recently updated from v1-0_stable dated a few weeks before. This problem existed with both versions. The phones are Cisco 7960's with SIP image 6.3, the calls are incoming on an E1 connected to a TE410P. Calls are queued and the phones statically defined as agents in the queue which has a ringall strategy. There is no queue announcement played to the answering user. Asterisk in vvv verbose doesn't log anything different when this occurs. So, has anyone else heard of something like this? Or have a suggestion for further investigation? Cheers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which Linux ?
Has anyone who has gotten Asterisk to run on the 2.6 kernel tested whether it is any faster/more-efficient at running Asterisk than 2.4? MATT--- -Original Message- From: Michael George [mailto:[EMAIL PROTECTED] Sent: Thursday, June 24, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 7:55 AM, Asterisk wrote: I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? Well, I will tell you, but I'll then have to kill you... Lucky for you I have no clue who you are or where you reside :) The incantation I used is: make linux26 That machine isn't running right now so I can't double-check it, but I don't recall having to do anything out of the ordinary. I do remember banging my head for a while before I remembered that there is a different build sequence for the 2.6 kernel... If this doesn't work for you, let me know and when I have a chance I'll fire that system up and give it another go-'round... - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:30 PM Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 2:33 AM, [EMAIL PROTECTED] wrote: From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. I asked a similar question about SuSE 9.1 and the 2.6 kernel a little while ago: http://lists.digium.com/pipermail/asterisk-users/2004-May/048781.html Follow the thread out and it seems some have had success. I have been doing our preliminary testbed with SuSE 9.0 and it is working just fine. Our final deployment will be under 9.1 and the 2.6 kernel. I already have * and zaptel built on that system, but we haven't run it yet. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk bypassed for name but not number - softphone
Greets, I have asteisk up and running, on asterisk console when i dial by extension (1) I see all transacitons of the call, but when I dial by name it seems to bypass the server. extensions.conf exten = 1,1,dial(SIP/option,20,tr) exten = option,1,goto(1,1) sip.conf [option] type=friend host=dynamic dtmf=inband username=option secret=option canreinvite=no callerid=Option 1 This * box is for sip softphone to softphone. Any ideas? __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager Commands - Timeout
Hi everyone, I am trying to generate calls from the asterisk manager. I am using perl and Socket.pm the Originate: Zap/g1/$phonenumber\r\n command I send works without any problem if I leave the Timeout: $timeout\r\n line out, however with the Timeout line in I get some very odd behaviour. I have searched the internet for ages and cannot find any useful information on what this command actually changes, if anyone would be kind enough to enlighten me on this subject It would be most appreciated. Ta. -- Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dead air on 7960 sip at start of call.
On Thu, Jun 24, 2004 at 01:45:55PM +0100, Chris Glover wrote: Hi, I've noticed the same thing with my 7960 running SIP 7.1. I came to the conclusion that the Cisco does some clever echo cancellation, and that pause is the echo canceller training. I don't think it's an Asterisk problem. My Grandstream 101 works perfectly. I had thought that it might be the echo canceller too... but of 400 calls today, I've had less than 20 reports of it occurring. Some only 2 seconds of silence, most around 5 to 6 seconds and a few over 20 seconds. I would have thought that's an awful long time for an echo canceller and fairly inconsistent. Or perhaps that's normal for echo cancellers? - Sam On Thu, 24 Jun 2004, Sam Tilders wrote: Hi, Despite much searching, I can't find anything quite like the problem we seem to be experiencing with our recently activated asterisk pbx. Of four people answering several calls per five minutes, I have reports of the occasional call starting with dead air. The user picks up the call from the queue, and they can hear that the line is too quiet. If they start talking, the caller either doesn't hear them, or only hears the end of the first sentence. Reports are that this silence at the start of the call lasts from 4-5 seconds, sometimes over 10 seconds and more rarely over 20 seconds. After the period of silence the call proceeds with no further problems. I have taken packet dumps of extended periods, there is nothing consistently different about the calls with silent start compared to those without the problem. The SIP INVITE is sent to the phone, the phone rings, and OK's the call. The RTP stream begins straight away. I'm using cvs-head as of 2004-06-10, CVS-D2004.06.09.14.00.00, recently updated from v1-0_stable dated a few weeks before. This problem existed with both versions. The phones are Cisco 7960's with SIP image 6.3, the calls are incoming on an E1 connected to a TE410P. Calls are queued and the phones statically defined as agents in the queue which has a ringall strategy. There is no queue announcement played to the answering user. Asterisk in vvv verbose doesn't log anything different when this occurs. So, has anyone else heard of something like this? Or have a suggestion for further investigation? Cheers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with PostgreSQL
Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql: got hostname of localhost Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:285 my_load_module: cdr_pgsql: got port of 5432 Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:288 my_load_module: cdr_pgsql: got user of asteriskpg Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:289 my_load_module: cdr_pgsql: got dbname of asteriskpgcdr Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:290 my_load_module: cdr_pgsql: got password of 65plesk Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Many thanks in advance Cheers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Manager Commands - Timeout
I use Net::Telnet in perl to interface the manager and it works much better. Take a look at the source code for some of the scripts in my astguiclient suite and you'll see all kinds of perl/Asterisk-manager data munging: http://astguiclient.sf.net/ MATT--- -Original Message- From: Luckcuck Nick-LCKN001 [mailto:[EMAIL PROTECTED] Sent: Thursday, June 24, 2004 8:57 AM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Asterisk Manager Commands - Timeout Hi everyone, I am trying to generate calls from the asterisk manager. I am using perl and Socket.pm the Originate: Zap/g1/$phonenumber\r\n command I send works without any problem if I leave the Timeout: $timeout\r\n line out, however with the Timeout line in I get some very odd behaviour. I have searched the internet for ages and cannot find any useful information on what this command actually changes, if anyone would be kind enough to enlighten me on this subject It would be most appreciated. Ta. -- Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
On Thu, 2004-06-24 at 08:10 -0400, Michael George wrote: On Jun 24, 2004, at 7:55 AM, Asterisk wrote: I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? Well, I will tell you, but I'll then have to kill you... Lucky for you I have no clue who you are or where you reside :) The incantation I used is: make linux26 That machine isn't running right now so I can't double-check it, but I don't recall having to do anything out of the ordinary. I do remember banging my head for a while before I remembered that there is a different build sequence for the 2.6 kernel... Just follow the notes in /usr/src/zaptel/README.Linux26 :) -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttool CLI
Matthew, Thanks a lot, that worked. Ivan Gostev [EMAIL PROTECTED] wrote: On Wed, Jun 23, 2004 at 12:45:59PM -0400, [EMAIL PROTECTED] wrote: I need to check red alarms status from the script, but asterisk CLI zap show channel 1 or pri show span 1 does not tell me this. zttool does, but I can run it only in interactive curses mode. Is there any ready solution? If you `cat /proc/zaptel/1` (substituting 1 with whatever span/card you would like to check, you should be able to see the alarm state of the card. Matthew Fredrickson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] How to force G729
If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to no, so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using disallow=all; allow=g729 in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I force it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says OK, now Dial(... @gateway), but force G729 (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? Now I better understand what you're trying to do. I'm not a programmer, but I'm fairly certain that you can't dynamically change codec preference within asterisk on a per call basis. However, just as soon as this gets posted, someone will likely jump all over that statement and post a way to do it. I don't think its and incoming vs outgoing issue. For each outgoing call, an rtp session is established between * and the gateway. That rtp session goes through a codec negotiation process that automatically selects a compatible codec based on what's common, and, when multiple choices are available, some other decision making process (transcode time, quality or something) that you probably don't have control over on a per call basis. So, my guess is you're not going to be able to do what you want. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 400SC and X100P
Thanks a lot for replying. I turned on the ACPI in the CMOS and it got better. At least I call receive several calls in sequence and call out but it hangs up right after the person gets the phone in the other side. So, something is still missing. What is your ACPI mode in the CMOS ? S1 or S3? Which kernel version are you using? Can you send me your .config ? Thanks again, Isamar On Thu, 24 Jun 2004, Martin List-Petersen wrote: Is your kernel ACPI enabled ? The motherboard in the PE400SC is basically the Dimension 8300, which i use for my development box with 1 X101P, 1 TDM400P and two ISDN cards here at home and that works without problems. One thing to make sure with these boards is that ACPI is enabled, since they are ACPI only. Kind regards, Martin List-Petersen On Thu, 2004-06-24 at 02:58, Isamar Maia wrote: I have a Dell PowerEdge 400SC with a X100P and a TDM01b. The board works wonderfully in another machine but in this brand new one, it just get in nuts. The problem is: 1) Zaptel recognizes it perfectly 2) No IRQ conflicts, two-wire new cable. 3) Asterisk starts up and listen the ring and answer the cal 4) RIght after answering the call, it's dropped. 5) The following calls, even with asterisk off, the driver(???) answers the call and hang it up. With the * running, it doesn't even get any ring, and the call is answered and dropped right away. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Really basic stuff :(
Gavin Hamill wrote: Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec etc.) goes directly to that machine. I am not doing any firewalling, nor is my ISP. I've made my configuration as superficial as I can to ease diagnosis: [EMAIL PROTECTED]:/etc/asterisk# ls -l -rw-r--r--1 root root 104 Jun 23 21:21 extensions.conf -rw-r--r--1 root root 164 Jun 23 19:25 iax.conf -rw-r--r--1 root root0 Jun 22 15:36 modem.conf -rw-r--r--1 root root 387 Jun 23 21:22 modules.conf -rw-r--r--1 root root 363 Jun 23 21:19 sip.conf -rw-r--r--1 root root0 Jun 22 15:36 voicemail.conf I know that this is not related to your ultimate question, but I would not recommend giving read access to everyone. Even if you have guest disabled, this still leaves you vulnerable to snoops discovering your configuration. With that in hand, they can make phone calls on your dime. I would change the access rights on all of these files to 640. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] How to force G729
Did you try having two sip.conf entries for your gateway? Forcing one with G729 and the other with ulaw? You would obviously need to change your dialplan accordingly and have each phone configured so that it would take the proper extension. I have not tried this, it is just really an idea... Manuel Wenger wrote: If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to no, so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using disallow=all; allow=g729 in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I force it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says OK, now Dial(... @gateway), but force G729 (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with PostgreSQL
On Thu, Jun 24, 2004 at 09:05:42PM +0800, Caleb Kow wrote: Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Asterisk is trying to connect to the postgresql postmaster. netstat -na |grep 5432 should show a line with LISTEN, if it does not then postgres is not accepting tcp connections. postmaster needs the -i option to accept tcp connections. The startup script, perhaps /etc/init.d/postgresql needs the command line to pg_ctl add the -i option to postmaster's command line. Something like: /usr/bin/pg_ctl -p /usr/bin/postmaster -o '-p 5432 -i' start with whatever other options are already there. Then after a restart, asterisk should be able to connect to postgres. -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk + appradius freeradius
I tried that module. First of all, you must be sure that your freeradius is working properly (just use radtest the usual way). If you had no problem compiling the module and intalling it via make install, then you must configure file /usr/local/etc/radius.conf (setting the right secret key). By the way, I have to move that file to directory /etc/asterisk. Finally, you must add an entry load = app_radius.so to your modules.conf file. I add an entry for test purposes in my radius users file (/etc/freeradius/users), just like this: Auth-Type := Local, User-Password == I assume the number must be the called-id associated with the channel you want to set a PIN behavior. You must also add an entry to the file extensions.conf like that: exten = s,1,Answer exten = s,2,SetLanguage(en); if you want to change language ... exten = s,3,Radius(CPP); CPP-Parameter is mandatory! exten = s,4,Hangup like documentation suggests (I don't really know what CPP means). That was the closer I get to the solution. Maybe is time for a Wiki page. On Wed, 23 Jun 2004, Harold Workman wrote: [EMAIL PROTECTED] wrote: Here is the jist: Freeradius is up running and functional using SIP Express radius how to. My asterisk box has app radius installed. Is there any documents on how-to link asterisk to freeradius? documentation is lacking on app radius, at least not as detailed as I need. Anyone know of a how-to or a link that covers this ? Thanks. Have you looked at http://appradius.minitelecom.org/ Doesnt look very detailed tho. Harold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roberto Paz The box said, 'Requires Windows 95 or better', so i installed Linux TKK 5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Video/H323/SIP
I tryed it. but callee cannot answering with video in SIP. # surely videosupport=yes in sip.conf H.323 is works well but I think stilln't support over * yet. mack_jpn. On Thu, 24 Jun 2004 14:03:10 +0200 Michael Devenijn [EMAIL PROTECTED] wrote: I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Linux ?
I did try that. Any help would be gratefully received. Julian - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 1:10 PM Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 7:55 AM, Asterisk wrote: I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? Well, I will tell you, but I'll then have to kill you... Lucky for you I have no clue who you are or where you reside :) The incantation I used is: make linux26 That machine isn't running right now so I can't double-check it, but I don't recall having to do anything out of the ordinary. I do remember banging my head for a while before I remembered that there is a different build sequence for the 2.6 kernel... If this doesn't work for you, let me know and when I have a chance I'll fire that system up and give it another go-'round... - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:30 PM Subject: Re: [Asterisk-Users] Which Linux ? On Jun 24, 2004, at 2:33 AM, [EMAIL PROTECTED] wrote: From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. I'm just about to try RedHat 8.0, which gets good comments in the wiki. I asked a similar question about SuSE 9.1 and the 2.6 kernel a little while ago: http://lists.digium.com/pipermail/asterisk-users/2004-May/048781.html Follow the thread out and it seems some have had success. I have been doing our preliminary testbed with SuSE 9.0 and it is working just fine. Our final deployment will be under 9.1 and the 2.6 kernel. I already have * and zaptel built on that system, but we haven't run it yet. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Did you try having two sip.conf entries for your gateway? Forcing one with G729 and the other with ulaw? You would obviously need to change your dialplan accordingly and have each phone configured so that it would take the proper extension. I have not tried this, it is just really an idea... That's actually a very good idea, and I have tried it: for outgoing calls it works like charm. But then the problem is transferred to incoming calls (from the gateway-asterisk-SIP client). Because the gateway now has 2 entries, asterisk is confused about what codec it has to use for incoming calls, and for some reason I can't force it, because the 2 entries have the same IP. I'm starting to think that I won't be able to solve that myself, but that someone will have to program something for this to work... But if I'm the only one having this kind of request, I'm not too optimistic -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Really basic stuff :(
On Thursday 24 June 2004 14:26, Stephen R. Besch wrote: Gavin Hamill wrote: I know that this is not related to your ultimate question, but I would not recommend giving read access to everyone. Even if you have guest disabled, this still leaves you vulnerable to snoops discovering your configuration. With that in hand, they can make phone calls on your dime. I would change the access rights on all of these files to 640. Thanks for the due diligence, but the way this box is configured and functioning right now, nobody's making any intelligible calls on it whatsoever, not even me =) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
I've just done the upgrade on my test asterisk system. On a series of outgoing PSTN calls via our TDM31B there was no detectable echo. I am curious did you play with the echotraining flag echotraining=yes or use the delay values for echotraining=some ms ? Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype 4 Linux
On Wed, 23 Jun 2004, Stefan de Konink wrote: Hi All, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation layer)). It is a bit of work, but reverse enginering is too :) Just write some code we can stick in asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Video/H323/SIP
Nakano San, Have you tried to make * only to route the connection and they just talk point-to-point without * bridging? Isamar On Thu, 24 Jun 2004, Masakazu Nakano wrote: I tryed it. but callee cannot answering with video in SIP. # surely videosupport=yes in sip.conf H.323 is works well but I think stilln't support over * yet. mack_jpn. On Thu, 24 Jun 2004 14:03:10 +0200 Michael Devenijn [EMAIL PROTECTED] wrote: I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with PostgreSQL
Caleb Kow wrote: Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql: got hostname of localhost Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:285 my_load_module: cdr_pgsql: got port of 5432 Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:288 my_load_module: cdr_pgsql: got user of asteriskpg Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:289 my_load_module: cdr_pgsql: got dbname of asteriskpgcdr Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:290 my_load_module: cdr_pgsql: got password of 65plesk Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Can you do a netstat -ap ? -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay in Zap Calls?
On Thu, 2004-06-24 at 03:35, Kannaiyan Natesan wrote:
I have this line in my extensions.conf,
exten = _393.,1,Dial(ZAP/${EXTEN:3},20,tr)
when I make a zap call, it gives me two rings and then makes the zap call.
Is there is a way I can make the call immediate?
Take off the r option. The r option tells Asterisk to provide a
ringing tone to the caller REGARDLESS of what the caller should be
hearing. You'll notice that if you call a busy number you'll hear a
ring or two and then the busy signal.
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Was this in the new samples from the CVS? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] How to force G729
Hmmm, I was thinking about this problem too... What type of gateway are you using? Is it registering with the Asterisk server? I would try using two different 'virtual' extensions on the gateway and in sip.conf. That way you would have full control on how calls from the gw to * are handled. Manuel Wenger wrote: That's actually a very good idea, and I have tried it: for outgoing calls it works like charm. But then the problem is transferred to incoming calls (from the gateway-asterisk-SIP client). Because the gateway now has 2 entries, asterisk is confused about what codec it has to use for incoming calls, and for some reason I can't force it, because the 2 entries have the same IP. I'm starting to think that I won't be able to solve that myself, but that someone will have to program something for this to work... But if I'm the only one having this kind of request, I'm not too optimistic -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
On Thursday 24 June 2004 09:01, Rich Adamson wrote: Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Just to be clear, to take advantage of the enhancement, do we need to pull the latest Zaptel, or Asterisk ... or both? Ryan Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Hmmm, I was thinking about this problem too... What type of gateway are you using? Is it registering with the Asterisk server? I would try using two different 'virtual' extensions on the gateway and in sip.conf. That way you would have full control on how calls from the gw to * are handled. I had thought about that, too ... Unfortunately the gateway is unable to register. We authenticate based on the IP address only. Otherwise, like you say, I could have 2 virtual extensions, but with IP only this is not possible. Maybe I will find a solution by sleeping over the problem (not physically, that is) tonight :-) -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Well there you go... No I hadn't done that. And it still sounded OK. Which is quite bizzare. Yesterday I has having terrible echo issues. Today none at all. *sigh* I'll see what setting echotraining=800 does for me... Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 12:21 AM, Chris Bond wrote: Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Was this in the new samples from the CVS? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Was this in the new samples from the CVS? Yes. I did a full Head checkout this morning, and the comments were included in the /usr/src/asterisk/configs/zapata.conf.samples (or something like that). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
They don't need to have the same IP. Assign several IP numbers to your linux box: ifconfig eth0:1 10.1.1.1 netmask 255.255.255.0 ifconfig eth0:2 10.1.1.2 netmask 255.255.255.0 Sorry guys... These are all great tips, but also this doesn't work: the gateway is not under my control, it is actually a real phone switch, which isn't owned by us. Unfortunately I can't tell them to add a second IP ... :-) -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
The change is only to chan_zap.c ... Since the changes were only to chan_zap.c, I'll have to guess it will improve the historic x100p echo issues as well, and ... So you should just need to pull the latest version of asterisk, depending on the age of your zaptel driver. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 1:35 AM, Ryan Courtnage wrote: On Thursday 24 June 2004 09:01, Rich Adamson wrote: Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Just to be clear, to take advantage of the enhancement, do we need to pull the latest Zaptel, or Asterisk ... or both? Ryan Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Ryan Courtnage wrote: Just to be clear, to take advantage of the enhancement, do we need to pull the latest Zaptel, or Asterisk ... or both? Anytime you update Asterisk you should always get a new Zaptel and libpri (if you are using it). I always do make clean install's everywhere too, just for a known sane build. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex
I am a new to asterisk, i wanted to test the opensource codec speex i have installed speex, and recompiled asterisk i can see the speex_codec.so getting loaded i have xten lite, i used the registry patch ( http://bugs.digium.com/bug_view_page.php?bug_id=918 ) but still when i use xten lite i get the following errors Jun 24 10:46:15 WARNING[-1305486416]: codec_speex.c:167 speextolin_framein: Out of buffer spaceJun 24 10:46:15 WARNING[-1305486416]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 512 frames am i doing something wrong? any pointers is helpful thanx sriram
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
I am curious did you play with the echotraining flag echotraining=yes or use the delay values for echotraining=some ms ? Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. I'd suggest that one may need to play with the delay to see what the tolerance is on your line, I am also interested as to why different line require different delay's b4 kicking in the echo can, is this due to the different switch that the line is connected to ? Rich, it would appear on your config the longer delay was the key to removing echo Just guessing here, I'd have to bet some money on the echo can being related to the CO switch. The reason for that bet is that historically (at least for last seven months or so), people either had echo or they didn't (like major 'yes' or no), and none of the research tracked against config parameters, motherboard model, pstn lines from a common source, type of lines, etc. Changing to 800 had such a significant impact that it's almost like someone purchased and installed an external echo can box. I don't think that I'd play around with other values though. There were three lines of code changed in chan_zap.c and two of those lines were dependent on values from each other. Example: changing from 400 (default) to 800 required another statement change from w to ww. I'd have to assume the w is a character/tone code to be used within the cancel training (or something like that). If you changed to echotraining=600, then how is the code going to insert a half a w? (I didn't look at the final cvs code, but would guess the choices are yes or 800, and everything else is silently ignored.) If you have echo, then try echotraining=800. If it disappears, don't fix something that's not broken. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
On Thursday 24 June 2004 09:01, Rich Adamson wrote: Per the doc in the configs samples, you have to implement echotraining=800 (instead of yes) to take advantage of the new code from yesterday. Just to be clear, to take advantage of the enhancement, do we need to pull the latest Zaptel, or Asterisk ... or both? For the echo problem, source code changes were made to only the /asterisk/channels/chan_zap.c file. Therefore, asterisk only. However, Mark fixed another problem associated with debouncing the ring indicator on the tdm card. That change was in zaptel/wcfxs.c and is unrelated to the echo problem. That bug caused asterisk to believe an incoming call was happening whenever any form of disturbance happened on the pstn line. (For example, with a bridged analog phone just taking the handset off-hook and replacing it would frequently cause * to believe the pstn line was ringing.) I'd pull updates for at least those two changes if your running in a stable production environment and don't want to be impacted by whatever other changes have gone into Head cvs. (For me, I just did a full checkout this morning and haven't been impacted (so far) by anything negative at all. But ours is a rather small soho system.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with PostgreSQL
Looks like PostgreSQL is running in UNIX local socket mode (which is default) and does not allow incoming TCP/IP connections even for localhost. Did you check for tcpip_socket = true line in your postgresql.conf file (it is /var/lib/pgsql/data/ directory on my system)? You can also check permissions in pg_hba.conf in the same directory. Ivan Gostev Caleb Kow wrote: Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql: got hostname of localhost Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:285 my_load_module: cdr_pgsql: got port of 5432 Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:288 my_load_module: cdr_pgsql: got user of asteriskpg Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:289 my_load_module: cdr_pgsql: got dbname of asteriskpgcdr Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:290 my_load_module: cdr_pgsql: got password of 65plesk Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Many thanks in advance Cheers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
On 25/06/2004, at 12:57 AM, Andrew Yager wrote: I'll see what setting echotraining=800 does for me... It still sounds good. There was no noticeable echo on the three calls we tried. Difficult to say whether it is a greater improvement or not, but I'm sure I'll have a feel for it after tomorrow. Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Use two separate entries with type=peer and type=user instead of one
entry with type=friend.
Tried that as well. This triggers yet another misbehaviour...
I tried to define 2 peers (for the outgoing calls), one called [gateway-g729] and one
called [gateway-ulaw], each allowing only the codec specified in the name. Then I
defined 1 user for incoming calls from the gateway (called [gateway-in]), with both
g729 and ulaw in the allow list.
And you know what happens? Outgoing calls are now fine (I can direct them either to
@gateway-g729 or @gateway-ulaw in the Dial() command), but incoming calls seem to have
a live on their own, and choose whatever codec they prefer. Even if I
setvar(SIP_CODEC=ulaw), the gateway-to-asterisk channel seems to remain in g729 (at
least that's what I can tell from show g729 - because sip show channels looks
correct, both ULAW).
At some point I get that message:
Jun 24 16:37:14 NOTICE[1104739248]: chan_sip.c:1314 sip_answer: Changing codec to
'ulaw' for this call because of ${SIP_CODEC) variable
And yes, in sip show channels the gateway-to-asterisk channel is marked as ULAW, but
for some reason a G729 license is used up, because the call did start in G729... Any
ideas?
I guess I'm very close to the solution, but now G729 licenses are acting weird and are
being used even in ULAW-to-ULAW calls which started with G729 in the beginning...
-Manuel
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Re: [Asterisk-Users] Delay in Zap Calls?
Thanks Eric. That works.
Kannaiyan
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 3:18 PM
Subject: Re: [Asterisk-Users] Delay in Zap Calls?
On Thu, 2004-06-24 at 03:35, Kannaiyan Natesan wrote:
I have this line in my extensions.conf,
exten = _393.,1,Dial(ZAP/${EXTEN:3},20,tr)
when I make a zap call, it gives me two rings and then makes the zap
call.
Is there is a way I can make the call immediate?
Take off the r option. The r option tells Asterisk to provide a
ringing tone to the caller REGARDLESS of what the caller should be
hearing. You'll notice that if you call a busy number you'll hear a
ring or two and then the busy signal.
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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[Asterisk-Users] Pulver's WiSIP with Linksys WAPs
I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP to maintain a reliable network connection with those WAPs. This is regardless of Linksys WAP11 firmware, WEP settings, distance from WAP, etc. While walking down a hallway, I did notice that the WiSIP did start working. Upon further investigation, the phone as I had it configured worked wonderfully with a Linksys WAP54g. Has anyone had similar problems? Anyone know a work around to get the WiSIP to work with Linksys WAP11's? We are planning on getting rid of all of our Linksys WAP's and replacing them with another vendors, but right now I have no idea if the WiSIP will or will not interoperate with the new WAPs. We have previously talked to Pulver Innovations on the phone, and the best they could offer was an RMA on the phone, but that was before I verified the phone functioned with the WAP54g. Since then we emailed them with additional information and have yet to hear back from them. -- William R. Thomas Corvar [EMAIL PROTECTED] Co-Webmaster http://www.theonering.net/ *#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#* I don't believe in sweeping social change being manifested by one person, unless he has an atomic weapon. -- Howard Chaykin If at first the idea is not absurd, then there is no hope for it. -- Albert Einstein X-Stamper-To: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P on a UK BT line ---- txgain issue
I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
---snip--- I don't think that I'd play around with other values though. There were three lines of code changed in chan_zap.c and two of those lines were dependent on values from each other. Example: changing from 400 (default) to 800 required another statement change from w to ww. I'd have to assume the w is a character/tone code to be used within the cancel training (or something like that). If you changed to echotraining=600, then how is the code going to insert a half a w? (I didn't look at the final cvs code, but would guess the choices are yes or 800, and everything else is silently ignored.) Don't see any change from w to ww in the CVS head if its the memset two lines above. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video/H323/SIP
Hi, -Original Message- Is there any software based solution to establish a video connection with * and sip protocol? MSN messenger 4.7 with any windows capturing device should work. Make sure you force the codecs properly, because MSN tries to negotiate some form of MJPEG which Asterisk doesn't support. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice ip10s
Hi, -Original Message- Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Ouch! Can you check if it is still fetching any config files from your FTP-server at boot ? Might be your configs are corrupted somehow. If it is not even doing that, you might just have to ship it back to SwissVoice and have them fix it :-P Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax with SPA-2000's?
On 2004.06.23 12:19 Lee Howard wrote: On 2004.06.18 23:15 Seth Mattinen wrote: I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO and going to something digital isn't an option. My setup looks like this: POTS --- X100P FXO -- Asterisk -- SPA-2000 SIP -- Fax I've got the same configuration, and it works fine, but I had to disable V.17 (i.e, 14400 baud). V.27, V.29, and *V.34* all seem work just fine, however. There seems to be some kind of incompatibility between Asterisk and V.17 (this isn't surprising, many PBXs screw up V.17 communications). I stand corrected. After a little bit of work with the fax application to adjust the timings (increasing all of the pauses), all is well with V.17 also. Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] How to force G729
On 07:01 AM 6/24/2004, Rich Adamson wrote: Now I better understand what you're trying to do. I'm not a programmer, but I'm fairly certain that you can't dynamically change codec preference within asterisk on a per call basis. However, just as soon as this gets posted, someone will likely jump all over that statement and post a way to do it. I don't think its and incoming vs outgoing issue. For each outgoing call, an rtp session is established between * and the gateway. That rtp session goes through a codec negotiation process that automatically selects a compatible codec based on what's common, and, when multiple choices are available, some other decision making process (transcode time, quality or something) that you probably don't have control over on a per call basis. So, my guess is you're not going to be able to do what you want. It sounds like what you are looking for is an Asterisk-wide (or perhaps channel-specific) preserve_codec option. Where preserve_codec=1 means that asterisk tries to preserve the originating codec if at all possible, and preserve_codec=0 lets asterisk freely choose any codec per whatever algorithm it chooses. As far as I know, this option doesn't exist, but depending upon the need, perhaps someone should issue a feature request. It seems like this might be an easy feature to add. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
I don't think that I'd play around with other values though. There were three lines of code changed in chan_zap.c and two of those lines were dependent on values from each other. Example: changing from 400 (default) to 800 required another statement change from w to ww. I'd have to assume the w is a character/tone code to be used within the cancel training (or something like that). If you changed to echotraining=600, then how is the code going to insert a half a w? (I didn't look at the final cvs code, but would guess the choices are yes or 800, and everything else is silently ignored.) Don't see any change from w to ww in the CVS head if its the memset two lines above. Yah I just checked that code to be sure but its ok to play with the delay that are *NO* depends on changing the delay, In fact the only issue you will have is a sanity check that echotraining=10-2000ms outside of that range you will get a warning and the value will default ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speex
Inband dtmf does not work with speex(only ulaw). Switch your dtmf mode to rfc2833. :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000 Sent: Thursday, June 24, 2004 11:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Speex I am a new to asterisk, i wanted to test the opensource codec speex i have installed speex, and recompiled asterisk i can see the speex_codec.so getting loaded i have xten lite, i used the registry patch ( http://bugs.digium.com/bug_view_page.php?bug_id=918 ) but still when i use xten lite i get the following errors Jun 24 10:46:15 WARNING[-1305486416]: codec_speex.c:167 speextolin_framein: Out of buffer space Jun 24 10:46:15 WARNING[-1305486416]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 512 frames am i doing something wrong? any pointers is helpful thanx sriram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile Error
Just did a new cvs download and then tried to compile. I get this error message: chan_zap.c:59:2: #error You need newer libpri Then there are some more chan_zap.c errors. Here is the cvs command: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel asterisk libpri And the make command #cd /usr/src/zaptel #make #cd /usr/src/asterisk #make And I did this after moving the current zaptel, asterisk, and libpri to archival. Where do I get this file? Or what am I doing wrong... -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Im having the same issue so far im on rxgain=2.0 and txgain=6.0. Seems to work perfectly apart from the echo issue. Im just about to checkout the latest cvs and apply the echotraining=800 Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I am having the same, some people can just about hear me while others do not say a thing or it is fine. I can hear them fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 24 June 2004 17:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X101P on a UK BT line txgain issue I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse
Still no go I have asked Digium tech support to look into it. I need the later cvs to get around a bug with the latest tdm400 card (load driver - unload driver - load driver again to make it work. t o n y On Thu, 2004-06-24 at 08:15, Tony Nichols wrote: On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pulver's WiSIP with Linksys WAPs
What type of configuration do you have setup with the phone? We found the units to be pretty unusable unless you set them up to use g729. With the g729 on 128 bit wep doesn't seem to quite have enough horse power, but 64 bit wep and no wep seem pretty workable. Before we tried the g729 we thought the phone was defective because it was so intermetent. We had similar if not identical problems with the wisips working better with better quality APs. Seems to work great with wap11s in our testing. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting William R Thomas [EMAIL PROTECTED]: I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP to maintain a reliable network connection with those WAPs. This is regardless of Linksys WAP11 firmware, WEP settings, distance from WAP, etc. While walking down a hallway, I did notice that the WiSIP did start working. Upon further investigation, the phone as I had it configured worked wonderfully with a Linksys WAP54g. Has anyone had similar problems? Anyone know a work around to get the WiSIP to work with Linksys WAP11's? We are planning on getting rid of all of our Linksys WAP's and replacing them with another vendors, but right now I have no idea if the WiSIP will or will not interoperate with the new WAPs. We have previously talked to Pulver Innovations on the phone, and the best they could offer was an RMA on the phone, but that was before I verified the phone functioned with the WAP54g. Since then we emailed them with additional information and have yet to hear back from them. -- William R. Thomas Corvar [EMAIL PROTECTED] Co-Webmaster http://www.theonering.net/ *#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#* I don't believe in sweeping social change being manifested by one person, unless he has an atomic weapon. -- Howard Chaykin If at first the idea is not absurd, then there is no hope for it. -- Albert Einstein X-Stamper-To: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
