Re: [Asterisk-Users] Netherlands zaptel.conf
I have TDM card with FXS on port 1, and the other ports are FXO's After isntalling the driver I execute the following commands: - modprobe wctdm - ztcfg -v ztcfg relpy: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. lsmod includes: wctdm 34592 0 zaptel 187012 1 wctdm zaptel.conf: fxoks=1 ;==1 FXS (phone device) fxsks=2-4 ;== 3 FXO's (phone lines) loadzone=nl defaultzone=nl Zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes ; define channels context=from-pstn ; Incoming calls go to [from-pstn] in extensions.conf signalling=fxs_ks ; Use FXS signalling for an FXO channel channel = 2-4 ; PSTN attached to port 2 3 4 context=to-pstn signalling=fxo_ks channel = 1 extensions.conf: [from-pstn] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Dial(SIP/200) exten = s,2,Hangup() Hope it helps Regards, Moutaz -- Original Message --- From: Pieter Claassen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tue, 16 May 2006 21:23:49 +0200 Subject: Re: [Asterisk-Users] Netherlands zaptel.conf On Tuesday 16 May 2006 20:39, M.Masri wrote: AFAIK UPC cable phone is a voip phone, and not PSTN Well, I tried to plug my KPN phone line into it as well with the same result. The PC refuses to answer using the fxsks protocol. I don't think these phone lines are IP carriers and suspect that UPC might turn the voice stream into something else in their modem. The phone however is a standard analogue device and I suspect you can stick anything you buy over the counter in there. I also swapped the FXO module into the first slot with no different results. Config included below. The question is how to start figuring out what is going on since I don't see any messages in /var/log/asterisk/* or syslog that indicates there is a problem? lsmod includes zaptel 225284 1 wcfxs /etc/zaptel.conf defaultzone=nl fxsks=1 loadzone=nl /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no context=incoming signalling=fxs_ks channel = 1 I am running ubuntu breezy and build the modules without much hastel from source. The only issue was it wanted gcc3.3 rather than gcc4 as a compiler and I had to remove the zaptel-source packages since it wrote a file that the zaptel-modules-2.6.10-5-386 wanted to overwrite. BTW. I also tried fxsls and fxsgs but nothing worked. I also received this error with fxsgs. Zaptel Configuration == Channel map: Channel 01: FXS Groundstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Any help appreciated. Pieter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netherlands zaptel.conf
Michiel van Baak wrote: I thought the wcfxs module is used for fxo cards? Anyhow, if I load the wcfxo module, then I get errors with ztcfg (below). ZT_CHANCONFIG failed on channel 1: No such device or address (6) Normally, if I load the wcfxs module and the zaptel module, then this is what I get (listed below). It all looks reasonable and the only issue seems to be that the card doens't answer the line. what port is the module in. If it is on port 3 or 4 then the zaptel and zapata needs to be the same. Why not just run the genzaptelconf file, or is that specific to AAH? Cheers Terry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
On 5/16/06, Edu [EMAIL PROTECTED] wrote: We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I Just for your info, I have experienced the same issue (just once) on a Dell PE 2850 also. Same error (10 retries, and on and on..) but the machine didn't freeze. Asterisk hung by that time, and had to be killed (-s 9), before we could start it again. After that, it ran good. It's been a week or 2 now, haven't had any issues yet. (* 1.2.7, zaptel 1.2.5, libpri 1.2.2) If you get any feedback, please share with the list.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
Hi All, We experienced this issue some time ago on our 2850. Question - Are you using queues ? Callum stoffell wrote: On 5/16/06, Edu [EMAIL PROTECTED] wrote: We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I Just for your info, I have experienced the same issue (just once) on a Dell PE 2850 also. Same error (10 retries, and on and on..) but the machine didn't freeze. Asterisk hung by that time, and had to be killed (-s 9), before we could start it again. After that, it ran good. It's been a week or 2 now, haven't had any issues yet. (* 1.2.7, zaptel 1.2.5, libpri 1.2.2) If you get any feedback, please share with the list.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
Or more specifically, are you using the AgentCallBackLogin function ? Callum McGillivray wrote: Hi All, We experienced this issue some time ago on our 2850. Question - Are you using queues ? Callum stoffell wrote: On 5/16/06, Edu [EMAIL PROTECTED] wrote: We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I Just for your info, I have experienced the same issue (just once) on a Dell PE 2850 also. Same error (10 retries, and on and on..) but the machine didn't freeze. Asterisk hung by that time, and had to be killed (-s 9), before we could start it again. After that, it ran good. It's been a week or 2 now, haven't had any issues yet. (* 1.2.7, zaptel 1.2.5, libpri 1.2.2) If you get any feedback, please share with the list.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netherlands zaptel.conf
Michiel van Baak wrote: If you load the wcfxs module and everything works (cept for the asterisk answering the phoneline) all is correct. wcfxs is for connecting an analog phone, not a PSTN connection. I think you have the wrong module on you wildcard to interface with the PSTN net. Sorry. Whoa, good call! I totally ignored that options. Pieter, what color is the module ? S110M = Green (for a phone device) X100M = Red (for a phone line) Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Having a Blonde moment.
On Tue, 16 May 2006, Hughes, Sam wrote: The context can be set when the agent(s) log in. AgentCallbackLogin([AgentNo|][Options|[EMAIL PROTECTED]) Many thanks, I knew I must be missing something simple. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EICON, chan_capi-cm and averlap receiving
Armin Schindler wrote: On Tue, 16 May 2006, Klaus Darilion wrote: Now I'm playing around with WaitExten , but I would prefer collecting digits in the channel. Why in the channel driver? The channel driver just provides the possibility for Asterisk to 'speak' CAPI (or any other API). The logic for what to do with the call and its information is core functionality and should be done by Asterisk. I think that is the reason why applications like WaitExten exists. Maybe you are right, and using WaitExten for sure gives you more control and increases flexibility. But on the other hand it also adds complexity to the dialplan. I like the overlap=yes feature as I do not have to care about collecting digits. regards Klaus PS: Probably I like to stay with old behaviors ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
We have a 2850 running with a Sangoma A102 , 50 agents using Agentcallbacklogin and around 4000 calls per day. No problems at all. With a te410 / 405 (we've got both, can't remember which one was in the dell) we had lockups almost every day. Julian. Klaus Darilion wrote: Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm and type of number problem (ToN)
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems with the ToN configurations in chan_capi-cm. I understand how
incoming calls are rewritten using national and international prefix. But for
outgoing calls - what is the ToN?
I never really needed ton in TE mode, but when your card is in NT mode,
setting the ton may be important.
Further, is there any debug info of the Ton? capi debug and the divactrl
dchannel both show the CLI and CDLI, but do not show the ToN.
The ton is a byte prepended to the callerid and can be seen when looking at
the capi messages. On outgoing call when 'capi debug' is set, the
presentation and ton value is shown as well.
Hi Armin!
I think the capi debug is too verbose. When I debug, I'm mostly
interested in D-channel info, not B-channel. What do you think about
splitting this into
capi debug dchannel and
capi debug
How can I set the ToN? I suspect chan_capi to use the received ToN also for
outgoing calls when bridging calls. How can I verify this?
When using a newer Asterisk with cid_ton (I think 1.2.x has them), then
chan-capi will set ton on incoming call and on outgoing call, this value is
used as well. So it is bridged.
This is IMO a strange thing, combined with the default behavior of
national and internationalprefix.
e.g. a call is received with CLI=4912345 TON=international
default internationalprefix=00
Thus, chan_capi rewrites the number to 004912345, cid_ton is still
international.
When this call is bridged to the PBX, it sends CLI=004912345
TON=international
Thus, if the number is rewritten, IMO also the context should be
rewritten to UNKNOWN.
Since Asterisk does not provide (as far as I know) a possibility to change
that cid_ton value, chan-capi will overwrite that value with the value in
variable ${CALLERTON}, if set.
As I see this must be a number. Using strings (INTERNATIONAL, LOCAL,
NATIONAL) would be nice too.
Not sure if I understand the source code right: CALLERTON will be read
to set callers TON on outgoing calls, whereas CALLEDTON will be set on
incoming called TON? Why not set both variables in incoming calls and
read both variables on outgoing calls?
regards
Klaus
When you use the latest chan-capi from trunk (HEAD version), you can use
'capi show channels' to see the ton as well.
Armin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm and dialing without number
Armin Schindler wrote: On Tue, 16 May 2006, Klaus Darilion wrote: Does someone have any hints? I would need a 'capi debug' log to say more, but chan-capi receives the command for dial these digits twice. Just a wild guess, but do you have softdtmf detection activated in addition? Yes - softdmtf is on! Thanks for the hint. btw: I use WaitExten to collect digits. Is softdtmf required for this application? softdtmf is needed when the hard-dtmf (DSP) is not available. If you use Eicon diva server cards, then softdtmf is never required. Thanks. I've no set softdtmf and relaxdtmf to off and things work fine now. regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] regexp
First, you can remove the quotes aorund your variable reference. I've seen examples with it, but you don't need it. I'm not sure: if variable is empty, you got an error. In addition, double quotes around text that may contain spaces will force the surrounded text to be evaluated as a single token. Second, I'm not sure what the tilde does after the equal sign, but asterisk won't understand it. What? ':' and '=~' are regexp operators in Asterisk. Regards DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] default password doesn't match
Hi all, This is my first post! I'm newbie, yesterday I installed [EMAIL PROTECTED], and Igot lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to connect via another computer in the same network, after trying to log - maint/password: FORBIDDEN You don't have permissions to access /main on this server Thanks in advance! Laura ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which is the best fax-modem for testing ?
2006/5/16, Rich Adamson [EMAIL PROTECTED]: So I was telling myself : what if I could buy the most inclusive fax-modem, connect it to a PC, and run a bunch of test scripts to gather useful information on both production and preparation systems ?. Total waste of money as the problem isn't the fax modem as noted above.Hi,In this case, the fax machine doesn't hangup (*) when connected to TDM400P FXS port.It seems related to electrical incompatibilities we couldn't remove with the help of Digium support though I can't personnally tell how far we really went into studying this case with them. You're certainly right in that electrical incompatibilities involve TDM400P capacities and Sangoma's A200D behaves differently.Reading past requests on this list, I saw people had fax machines working with TDM400P. So there must be something somewhere explaining why it doesn't work in my case.I thought a super fax-modem could be used as a reference case : you send faxes with as many different settings as possible (speeds, protocols, flash signals levels, ...) and then analyse performances. Regards(*) By fax doesn't hangup, I mean though Asterisk server forward an incoming fax call to the right extension, it keeps on ringing the fax machine which never hangup. Maybe the flash signal is too weak ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no SUBSCRIBE request sent
Hi all, i am playing around with several optipoint4x0 and run into trouble trying to get hint functionality to work. I notice that there is no status notifications. But afaik this should be implemented via the SUBSCRIBE/NOTIFY mechanism. I can see INVITE, TRYING, RINGING, ACK, BYE but no SUBSBCRIBE in my sip debug traces. I have problem to understand how hint priority works. I follow the instructions from http://www.voip-info.org/wiki/index.php page=Asterisk+standard+extensions but it still doesn't work. [2001] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [2002] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [local] exten = 2001,1,Dial(SIP/2001,10,tr) exten = 2002,1,Dial(SIP/2002,10,tr) [notify] exten = 2001,hint,SIP/2001 exten = 2002,hint,SIP/2002 thx in advance for your help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no SUBSCRIBE request sent
On 17/05/2006, at 7:36 PM, richard Coco wrote: [local] exten = 2001,1,Dial(SIP/2001,10,tr) exten = 2002,1,Dial(SIP/2002,10,tr) [notify] exten = 2001,hint,SIP/2001 exten = 2002,hint,SIP/2002 Try this: [local] exten = 2001,1,Dial(SIP/2001,10,tr) exten = 2001,hint,SIP/2001 exten = 2002,1,Dial(SIP/2002,10,tr) exten = 2002,hint,SIP/2002 cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NO ringing tone while dialing
Hi everybody,
I don't know how to do this,
I redirect a call and then dial someone, but I dont want the ringing
tone to be listened while the dialing part is waiting for the dialed
part to pick up the phone.
exten = 555,2,Dial(${STRING4},30)
I have tried when the option 'm' , but I don´t want the default music
on hold to be listened neither. I want nothing (silence) to be heard
instead of ring, ring.
Any idea how to do this???
Thanks in advance,-- María
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Min-Expires
I am trying to register my Asterisk server to a SIP server which doesn't accept an Expires: field smaller than 1800 seconds and indicates it correctly with a Min-Expires: in an error response when Asterisk tries to use its default of 120 seconds. Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the default_expirey globally. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netherlands zaptel.conf
On Wednesday 17 May 2006 08:48, Florian Overkamp wrote: Michiel van Baak wrote: If you load the wcfxs module and everything works (cept for the asterisk answering the phoneline) all is correct. wcfxs is for connecting an analog phone, not a PSTN connection. I think you have the wrong module on you wildcard to interface with the PSTN net. Sorry. Whoa, good call! I totally ignored that options. Pieter, what color is the module ? S110M = Green (for a phone device) X100M = Red (for a phone line) The module is red and plugged into the UPC line (phone line). I also moved the module from the 4th slot on the card to the first slot. I plugged a power source from the motherboard into the card (not sure if it is needed). So, it looks like I have the right module installed. Is the wcfxs the correct module to load for a fxo interface? Any further comments appreciated. Pieter [EMAIL PROTECTED]:~ # lsmod |grep zap zaptel225284 1 wcfxs crc_ccitt 2176 2 hisax,zaptel [EMAIL PROTECTED]:~ # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Registers
On 05/17/06 04:00 Noah Miller said the following: only one registration. You can register from multiple devices, but only the one that has most recently registered will receive calls. Put another way, when the second device registers it will unregister the first device. exactly as you've put it for incoming calls. however, in practice, both devices will be able to make outgoing calls. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no SUBSCRIBE request sent
Hi, first of all, sorry for this long thread... I have changed my extensions.conf like you suggested and delete the line with subscribecontext=notify. But unfortunately i still don't see subscribe request in the sip debug trace. SIP Debugging enabled kingcoco*CLI -- SIP read from 192.168.204.5:6108: --- (0 headers 0 lines) Nat keepalive --- kingcoco*CLI -- SIP read from 192.168.204.100:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 307 Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983 Call-ID: 7e6c264483fd010 From: OptiPoint410std sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED] CSeq: 1 INVITE Supported: timer Min-SE: 90 Supported: 100rel Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Content-Type: application/sdp Contact: OptiPoint410std sip:[EMAIL PROTECTED]:5060;transport=udp Supported: replaces User-Agent: optiPoint 410_420/v4 4.1.66 v=0 o=MxSIP 0 1595508908 IN IP4 192.168.204.100 s=SIP Call c=IN IP4 192.168.204.100 t=0 0 m=audio 5004 RTP/AVP 9 8 0 18 4 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (17 headers 14 lines)--- Using INVITE request as basis request - 7e6c264483fd010 Sending to 192.168.204.100 : 5060 (non-NAT) Found user '2001' Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.204.100:5004 Found description format G722 Found description format PCMA Found description format PCMU Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2002 in local (domain 192.168.204.223) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=udp Transmitting (no NAT) to 192.168.204.100:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100 From: OptiPoint410std sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED] Call-ID: 7e6c264483fd010 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Dial(SIP/2001-65fe, SIP/2002|10|tr) in new stack We're at 192.168.204.223 port 10830 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.204.5:6108: INVITE sip:[EMAIL PROTECTED]:6108 SIP/2.0 Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport From: OptiPoint410std sip:[EMAIL PROTECTED];tag=as29a3f9ee To: sip:[EMAIL PROTECTED]:6108 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 May 2006 08:58:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 246 v=0 o=root 24071 24071 IN IP4 192.168.204.223 s=session c=IN IP4 192.168.204.223 t=0 0 m=audio 10830 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2002 Transmitting (no NAT) to 192.168.204.100:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100 From: OptiPoint410std sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED];tag=as5094780f Call-ID: 7e6c264483fd010 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- kingcoco*CLI -- SIP read from 192.168.204.5:6108: SIP/2.0 180 Ringing To: sip:[EMAIL PROTECTED]:6108;tag=0a630b27 From: OptiPoint410stdsip:[EMAIL PROTECTED];tag=as29a3f9ee Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:6108 Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/2002-7bc1 is ringing kingcoco*CLI -- SIP read from 192.168.204.5:6108: SIP/2.0 200 OK To: sip:[EMAIL PROTECTED]:6108;tag=0a630b27 From: OptiPoint410stdsip:[EMAIL PROTECTED];tag=as29a3f9ee Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:6108 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 185 v=0 o=- 10603328
[Asterisk-Users] (newbie) Zaptel/ztdummy compiling on debian
Im trying to compile Zaptel driver with the ztdummy. I have no hardware cards from digium. I tried following steps: http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy Im running : Linux version 2.4.27-2-386 ([EMAIL PROTECTED]) (gcc version 3.3.5 (Debian 1:3.3.5-13)) #1 Wed Aug 17 09:33:35 UTC 2005 And Asterisk : 1.2.7.1 And this is what I get :s Does anybody have an idea what is wrong. Prob. something stupid I guess ASTERISK:/usr/src/zaptel-1.2.5# modprobe ztdummy /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rf7663209 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol add_wait_queue_R2cea9688 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R31ed257b /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_wait_queue_Ree3648ba /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol __pollwait_R35631129 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol create_proc_entry_R648035a2 /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod /lib/modules/2.4.27-2-386/misc/zaptel.o failed /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod ztdummy failed WISMA-ASTERISK:/usr/src/zaptel-1.2.5# modinfo ztdummy filename: /lib/modules/2.4.27-2-386/misc/ztdummy.o description: Dummy Zaptel Driver author: Robert Pleh [EMAIL PROTECTED] license: GPL parm: debug int parm: monitor int ASTERISK:/usr/src/zaptel-1.2.5# modinfo usb-uhci filename: /lib/modules/2.4.27-2-386/kernel/drivers/usb/host/usb-uhci.o description: USB Universal Host Controller Interface driver author: Georg Acher, Deti Fliegl, Thomas Sailer, Roman Weissgaerber license: GPL ASTERISK:/usr/src/zaptel-1.2.5# depmod -ae depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: add_wait_queue_R2cea9688 depmod: remove_wait_queue_Ree3648ba depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: proc_mkdir_Rf7663209 depmod: add_wait_queue_R2cea9688 depmod: remove_proc_entry_R31ed257b depmod: remove_wait_queue_Ree3648ba depmod: __pollwait_R35631129 depmod: create_proc_entry_R648035a2 depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/ztd-eth.o depmod: dev_add_pack_R695fe0c9 depmod: skb_under_panic_Rd7c1ee8c depmod: dev_get_by_name_R114d4d81 depmod: dev_remove_pack_Raf74dcbe depmod: __kfree_skb_R5c3bf84d depmod: skb_over_panic_R635aef7c depmod: dev_queue_xmit_Rc28f17b6 depmod: alloc_skb_Ra26ebbf6 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Plan to free myself from AAH
Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to rewrite dialplan from scratch and I'd like only to clean actual AAH dialplan. I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A CDR issue of agent.conf createlink feature
Hi,
Asterisk version : 1.2.7.1 stable version
We try agent.conf setting of
createlink=yes
We always can not see this link value to be filled in MySQL's
table filed : userfield
But we can see the record file has been created correctly.
In debug mode, no userfiled shown in SQL command,
May 17 18:10:51 DEBUG[2889] pbx.c: Function result is
'"unknown" 2001'May 17 18:10:51 DEBUG[2889] pbx.c: Function result
is '2001'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is '1'May 17
18:10:51 DEBUG[2889] pbx.c: Function result is 'sip_ps'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'SIP/2001-783e'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'Agent/1000'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'Queue'May 17 18:10:51 DEBUG[2889]
pbx.c: Function result is '1180'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:40'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:41'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:51'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '11'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is
'10'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is 'ANSWERED'May
17 18:10:51 DEBUG[2889] pbx.c: Function result is 'DOCUMENTATION'May 17
18:10:51 DEBUG[2889] pbx.c: Function result is '2001'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is '1147860640.0'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'agent-1000-1147860640-3.wav'May 17
18:10:51 DEBUG[2889] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.May 17 18:10:51 DEBUG[2889] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2006-05-17 18:10:40','\"unknown\" 2001','2001','1','sip_ps',
'SIP/2001-783e','Agent/1000','Queue','1180',11,10,'ANSWERED',3,'2001')May
17 18:10:51 DEBUG[2889] chan_sip.c: update_call_counter(2001) - decrement call
limit counter
Do I miss any important flag in config to enable this
field?
best regard
kaiser
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm and type of number problem (ToN)
On Wed, 17 May 2006, Klaus Darilion wrote:
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems with the ToN configurations in chan_capi-cm. I
understand how
incoming calls are rewritten using national and international prefix.
But for
outgoing calls - what is the ToN?
I never really needed ton in TE mode, but when your card is in NT mode,
setting the ton may be important.
Further, is there any debug info of the Ton? capi debug and the
divactrl
dchannel both show the CLI and CDLI, but do not show the ToN.
The ton is a byte prepended to the callerid and can be seen when looking
at the capi messages. On outgoing call when 'capi debug' is set, the
presentation and ton value is shown as well.
Hi Armin!
I think the capi debug is too verbose. When I debug, I'm mostly interested in
D-channel info, not B-channel. What do you think about splitting this into
capi debug dchannel and
capi debug
on 'capi debug' you see messages for your set verbose level only.
With 'set verbose 5' you see the 'dchannel' stuff, with 'set verbose 9'
you see everything.
How can I set the ToN? I suspect chan_capi to use the received ToN
also for
outgoing calls when bridging calls. How can I verify this?
When using a newer Asterisk with cid_ton (I think 1.2.x has them), then
chan-capi will set ton on incoming call and on outgoing call, this value
is used as well. So it is bridged.
This is IMO a strange thing, combined with the default behavior of national
and internationalprefix.
e.g. a call is received with CLI=4912345 TON=international
default internationalprefix=00
Yes, but that can be changed in capi.conf.
Thus, chan_capi rewrites the number to 004912345, cid_ton is still
international.
When this call is bridged to the PBX, it sends CLI=004912345 TON=international
Thus, if the number is rewritten, IMO also the context should be rewritten to
UNKNOWN.
The user has two possibilities:
a) use the chan-capi feature of automatically prepend the prefixes according to
TON
b) don't use automatic prefix setting in capi.conf and do your own stuff
according to CALLERTON in your dialplan
both should not be mixed, because (as you stated above) will cause double
changes. a) is for standard usage (one ISDN port, no NT-mode), b) is the
professional version.
Since Asterisk does not provide (as far as I know) a possibility to
change that cid_ton value, chan-capi will overwrite that value with the
value in
variable ${CALLERTON}, if set.
As I see this must be a number. Using strings (INTERNATIONAL, LOCAL, NATIONAL)
would be nice too.
Hmm, you can device variables with that in your dialplan. Or Asterisk
provides them...
Not sure if I understand the source code right: CALLERTON will be read to set
callers TON on outgoing calls, whereas CALLEDTON will be set on incoming
called TON? Why not set both variables in incoming calls and read both
variables on outgoing calls?
CALLEDTON is set by chan-capi for the TON of the called number, not the
caller id! Normaly you don't need that TON and Asterisk doesn't provide
any feature for that.
The CALLERTON is the widely used TON and it is set to asterisks cid_ton
internal variable, which can be read via ${CALLERTON}. But since Asterisk
does not set cid_ton when writing to CALLERTON, chan-capi evaluates this
variable by itself (when set).
Armin
regards
Klaus
When you use the latest chan-capi from trunk (HEAD version), you can use
'capi show channels' to see the ton as well.
Armin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] default password doesn't match
Hi, I suspect that either the permissions are wrong for /main or there are no files in it and directory listings are denied. It sounds like an incomplete install to me. try and ssh onto it and do an ls -lh in /main and see if there are any files in there Alasdair Laura Barquín wrote: Hi all, This is my first post! I'm newbie, yesterday I installed [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], and I got lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to connect via another computer in the same network, after trying to log - maint/password: FORBIDDEN You don't have permissions to access /main on this server Thanks in advance! Laura ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Alasdair Gow BSc (Hons) Support Specialist Colloquium Internet Support ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no SUBSCRIBE request sent
On 17/05/2006, at 8:27 PM, richard Coco wrote: unfortunately i still don't see subscribe request in the sip debug trace. Have you configured your phone to subscribe to the extension? :) cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netherlands zaptel.conf
On Wed, May 17, 2006 at 08:29:14AM +0200, Terry Wade wrote: Why not just run the genzaptelconf file, or is that specific to AAH? genzaptelconf is not specific to AAH. Not originally from there, actually. A recent version of it could be found in latest debian packages and Xorcom Rapid packages. Not to mention the trunk of zaptel (under xpp) -- Tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plan to free myself from AAH
On 12:30, Wed 17 May 06, Mimmus wrote: Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to rewrite dialplan from scratch and I'd like only to clean actual AAH dialplan. I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Looks fine to me -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Deadlocks in 1.2.7.1
Hello! Unfortunately we are seeing lately (2-3 times during a day) that asterisk seems to hang up somehow - no new calls can be made and sip show peers and other commands show no obvious problem. We then recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile and now we see the following messages: May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock? waited 460 sec for mutex 'iflock'? May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): 'iflock' was locked here. May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 __ast_pthread_mutex_lock: pbx.c line 2017 (ast_extension_state_del): Deadlock? waited 460 sec for mutex 'hintlock'? May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): 'hintlock' was locked here. May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock? waited 460 sec for mutex 'iflock'? May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): 'iflock' was locked here. May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 __ast_pthread_mutex_lock: pbx.c line 2017 (ast_extension_state_del): Deadlock? waited 460 sec for mutex 'hintlock'? May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): 'hintlock' was locked here. May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock? waited 460 sec for mutex 'iflock'? May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): 'iflock' was locked here. May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 __ast_pthread_mutex_lock: pbx.c line 2017 (ast_extension_state_del): Deadlock? waited 460 sec for mutex 'hintlock'? May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): 'hintlock' was locked here. May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock? waited 460 sec for mutex 'iflock'? May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): 'iflock' was locked here. This continues until someone stops asterisks and restarts it. I googled around and found some asterisk deadlock problems but not refering to iflock or hintlock. I glanced through the source looking for mutex_(un)locks iflock and found them in pairs, however sometimes some a screen page apart. We dont use any hardware in the machine and the partners for the asterisk server are a couple of VoiP phones, a handful sub asterisk on other machines and a Cisco 5300 for the PSTN gateway. Philosophy: Dont you want to get rid of the threads and therefore mutex-problems? I once wrote a sort-of-asterisk for NMS AG cards and used threads as well and saw similar problems, and since I rewrote the whole project to a single threaded event-state machine it never crashed or hung anymore aside from 0-strcpy-memory access :-) Second I remember that malloc() in 2 different threads (kernel 2.4.20) returned the same memory pointer - so I had to encapsulate malloc() with a mutex_lock/unlock too. Any clues why this possible deadlock happens? Regards Philipp Ott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager and Events Problem
Sometimes (once per 4000 lines or so - depending on speed of network) manager improperly returns events. For example, one QueueMember will get overwritten by (or as part of) another, like this (see third line): Event: QueueMember Queue: 09 LocatiEvent: QueueMember Queue: 09 Location: Agent/09003 Membership: static Penalty: 1 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 The details were reported at http://bugs.digium.com/view.php?id=7116. In short: this behaviour can easily be reproduced in branch 25988, or releases 1.2.x (including 1.2.7.1), by repeatedly sending QueueStatus commands. However, I have not been able to reproduce this in trunk 25930. My guess is that there is something buggy in the locking mechanism of the current Asterisk releases. What would be the fastest way to solve this problem, as this is causing me lots of problems in my production system? Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netherlands zaptel.conf
The correct module to load for a TDM Cards interface is wctdm see http://www.digium.com/en/docs/misc/quick_install_zaptel_asterisk.pdf Regards, Moutaz -- Original Message --- From: Pieter Claassen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Sent: Wed, 17 May 2006 12:17:56 +0200 Subject: Re: [Asterisk-Users] Netherlands zaptel.conf On Wednesday 17 May 2006 08:48, Florian Overkamp wrote: Michiel van Baak wrote: If you load the wcfxs module and everything works (cept for the asterisk answering the phoneline) all is correct. wcfxs is for connecting an analog phone, not a PSTN connection. I think you have the wrong module on you wildcard to interface with the PSTN net. Sorry. Whoa, good call! I totally ignored that options. Pieter, what color is the module ? S110M = Green (for a phone device) X100M = Red (for a phone line) The module is red and plugged into the UPC line (phone line). I also moved the module from the 4th slot on the card to the first slot. I plugged a power source from the motherboard into the card (not sure if it is needed). So, it looks like I have the right module installed. Is the wcfxs the correct module to load for a fxo interface? Any further comments appreciated. Pieter [EMAIL PROTECTED]:~ # lsmod |grep zap zaptel225284 1 wcfxs crc_ccitt 2176 2 hisax,zaptel [EMAIL PROTECTED]:~ # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Reasons for a SIP channel to hang ? - partially resolved
Ok, I got rtptimeout setup to 60 in sip.conf and it go better ; came back to normal as soon as I put it. If anybody knows if rtpkeepalive and rtptimeout can work in conjunction, please share your toughts ! Thanks, Fred - Original Message - From: Frederic Jean To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 16, 2006 10:51 Subject: Reasons for a SIP channel to hang ? Hi all, Simple question; What are the possible reasons for a SIP channel to hang there for hours after the call has terminated ? Calls are sent from 1.2.6 to SER using DeadAGI over the internet, and yesterday there were 100 calls that hanged for hours over a total of 15k calls. Typical day would be 3 to 5 calls, I also noticed that a reboot helps to resolve this issue. Thanks, Fred ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no SUBSCRIBE request sent
Hi again, what do you mean exactely with Have you configured your phone to subscribe to the extension? :). I have several optipoint410 and eyebeam. On one of the Optipoint(exten 2001) i have configured a selected dialing bottum with the extensions of the eyebeam(exten 2002). Do i need more configuration on the IP-phone? thx in advance --- Avi Miller [EMAIL PROTECTED] wrote: On 17/05/2006, at 8:27 PM, richard Coco wrote: unfortunately i still don't see subscribe request in the sip debug trace. Have you configured your phone to subscribe to the extension? :) cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
Callum McGillivray schrieb: We experienced this issue some time ago on our 2850. Question - Are you using queues ? In my case we did not use queues and we had the problems with a custom made machine with a Intel Torrey Pines Mainboard. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200D problem
I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. Well, just to complete my own thread, this seems to be a probable hardware defect and Sangoma is sending a replacement. I had to live with software echo cancellation for a day or two - shudder - it's amazing how much better the hardware echo cancellation is! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which is the best fax-modem for testing ?
So I was telling myself : what if I could buy the most inclusive fax-modem, connect it to a PC, and run a bunch of test scripts to gather useful information on both production and preparation systems ?. Total waste of money as the problem isn't the fax modem as noted above. Hi, In this case, the fax machine doesn't hangup (*) when connected to TDM400P FXS port. It seems related to electrical incompatibilities we couldn't remove with the help of Digium support though I can't personnally tell how far we really went into studying this case with them. Are you in the US? You might try using a plain old voltmeter on tip ring to see if there is any form of disconnect signal. In the US, you should see the voltmeter going to zero volts for at least a 1/4 second or so. If you don't see the disconnect, then the problem is asterisk/tdm oriented. If you do see the disconnect, the problem is in the fax machine. You're certainly right in that electrical incompatibilities involve TDM400P capacities and Sangoma's A200D behaves differently. Reading past requests on this list, I saw people had fax machines working with TDM400P. Some folks have been able to make it work, but its a very small percentage of implementations. The general consensus is that if zttest reports anything less then about 99%, faxes will not work properly. The problem seems to be oriented around missed/lost data frames (across the pci bus) every xx number of seconds using the TDM card. If the missed/lost data occurs when the fax modem is actually sending data, the reproduced analog signal will be distorted. If it occurs between bursts of fax data, its less impacting. Its similar to clock slippage where clocking regains sync after xx seconds. The same missed/lost data frames occur with the A200D, negatively impacting fax modem usage if the fax call crosses the pci bus. However, if the fax call stays on the A200D (as in fxs - fxo on the exact same card), faxes function very reliably. So there must be something somewhere explaining why it doesn't work in my case. I thought a super fax-modem could be used as a reference case : you send faxes with as many different settings as possible (speeds, protocols, flash signals levels, ...) and then analyse performances. That might provide some insight into the issue, but I don't believe its going to provide much in terms of root cause. Regards (*) By fax doesn't hangup, I mean though Asterisk server forward an incoming fax call to the right extension, it keeps on ringing the fax machine which never hangup. Maybe the flash signal is too weak I'm very confused by the above statement. What do you mean by it keeps on ringing and machine never hangup in the same sentence? (No such thing as it keeps ringing and never hangup. Hangup occurs after answering, so if its ringing, it can't hangup.) What do you mean by flash signal is too weak? (There's no such thing as a weak flash. Sort of equivalent to saying a weak binary 1.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM does not disconnect
Hello all.
This is my very first message to the list. I have a TDM400P card, It
has 2 FXO channels which are connected to extensions of my PBX
(Ericsson BP250), so I can dial from any SIP softphone directly to
physical (analog and digital) extensions on my company.
My PBX is configured so when I dial 8 on any extension, it will
redirect to the first free FXO channel on my TDM400P card. Then I use
the Asterisk's DISA application to get a dial tone, like this:
exten = s,1,disa(no-password,tdm-disa)
[tdm-disa]
exten = _XXX.,1,ChanIsAvail(Zap/3Zap/4) ; Checks for a free channel to dial
exten = _XXX.,2,Dial(${AVAILORIGCHAN}/${EXTEN}) ; Dials the number on
the first channel available
But if the person I'm calling does not answer the phone and I hangup
(fisically) the extension, the Zap channels doesn't hangup! They stay
connected, and the line I called keeps on ringing.
So, this is the entire process:
1. I pickup a physical extension, and dial 8
2. The PBX redirects the call to the first FXO channel available
3. Asterisk answers the call and gives a dial tone using the DISA application
4. I dial the number I want
5. Asterisk dials using an available Zap channel
6. If the person I called does not answer the phone, I hangup my
extension but the FXO channels doesn't hangup!
This is the logs I got running asterisk -vvv on the
situation above. My comments on it are rounded with []:
[I pickup my physical extension and dial 8]
-- Starting simple switch on 'Zap/3-1'
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18
(Ring Begin)...
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 2
(Ring/Answered)...
May 17 08:48:56 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18
(Ring Begin)...
-- Executing DISA(Zap/3-1, no-password|tdm-disa) in new stack
[Asterisk gives me dial tone and I dial 081168345 - 0 + my cell phone number]
-- Executing ChanIsAvail(Zap/3-1, Zap/3Zap/4) in new stack
-- Hungup 'Zap/4-1'
-- Executing NoOp(Zap/3-1, Canal: Zap/4) in new stack
-- Executing Dial(Zap/3-1, Zap/4/081168345) in new stack
-- Called 4/081168345
[My cell phone starts to ring, I hangup my extension. Cell phone keeps
on ringing.]
[After a while (about one minute) the following shows up]
-- Zap/4-1 is busy
-- Hungup 'Zap/4-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Hungup 'Zap/3-1'
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
Hello Julian! As we are using the same HW (2850 and Sangoma cards) but have some problems with AMI (* manager interface) I wonder which OS (version of Linux, kernel version) are you using? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P We have a 2850 running with a Sangoma A102 , 50 agents using Agentcallbacklogin and around 4000 calls per day. No problems at all. With a te410 / 405 (we've got both, can't remember which one was in the dell) we had lockups almost every day. Julian. Klaus Darilion wrote: Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: attended transfer issue
Hi, Following last thread onunifying blind and attendedtransfers (http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/146002/focus=146683) I think it would be great if a user could either : 1. transform a transfer into into a blind-and-forget transfer one pressing # key 2.transformthat transfer into a blind-and-return pressing another key As its name suggests, a blind-and-return transfer is a transfer which return back to initial callee in case transfer calleedoesn't answer (useful for receptionnist who can't attend nor loose calls to special extensions). What do you think of that ? Maybewe could enchance http://bugs.digium.com/view.php?id=6973description though a bounty on voip-info.org could be the best place to act on. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
What problems are you having ? We are using CentOS 4.2 Linux 2.6.9-22.0.1.ELsmp Asterisk SVN-trunk-r7353M I know, time to update, but I am not in the office currently and do *not* want to do it remotely ;) Julian Asterisk wrote: Hello Julian! As we are using the same HW (2850 and Sangoma cards) but have some problems with AMI (* manager interface) I wonder which OS (version of Linux, kernel version) are you using? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P We have a 2850 running with a Sangoma A102 , 50 agents using Agentcallbacklogin and around 4000 calls per day. No problems at all. With a te410 / 405 (we've got both, can't remember which one was in the dell) we had lockups almost every day. Julian. Klaus Darilion wrote: Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_perl voor asterisk 1.2.4
Title: Running commands from dialplans Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE=\/usr/local/res_perl\ -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE - D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm - I/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE - I/usr/src/bristuff-0.3.0-PRE-1l/asterisk-1.2.4/ - I/usr/src/bristuff-0.3.0-PRE-1l/asterisk-1.2.4//include -I. -c AstAPIBase.c AstAPIBase.c: In function `asterisk_recordfile': AstAPIBase.c:435: warning: ISO C90 forbids mixed declarations and code AstAPIBase.c: In function `asterisk_request_and_dial': AstAPIBase.c:813: warning: passing arg 6 of `ast_request_and_dial' makes integer from pointer without a cast AstAPIBase.c:813: error: too few arguments to function `ast_request_and_dial' AstAPIBase.c: In function `asterisk_request': AstAPIBase.c:880: error: too few arguments to function `ast_request' make: *** [AstAPIBase.o] Error 1 Can anybody tell me if this version is the right res_perl version? Kind regards. Arjan Kroon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay when ringing internal extensions on incoming zap call
Going to AMP, Setup - General - Extension of fax machine for receiving faxes = disabled *should* disable fax detection by causing it to use a different branch of the AMP macro's... I did set it to disabled, but it still called NVFaxDetect() with a parameter of zero. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
We have problems with 'loosing' parts of messages sent from Asterisk Manager to our CTI Server (we also tested it with another test program, but the problem persisted). Sometimes (once per 4000 lines or so - depending on speed of network) manager improperly returns events. For example, one QueueMember will get overwritten by (or as part of) another, like this (see third line): Event: QueueMember Queue: 09 LocatiEvent: QueueMember Queue: 09 Location: Agent/09003 Membership: static Penalty: 1 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 We are using Red Hat ES Linux 4 (the same 2.6.9-22 kernel) but SVN-branch-1.2-r27093M. We also noticed that problem seems to be fixed in trunk, but we are not 'brave' enough to use trunk in production environment. But obviously your experiences with using trunk in production are quite good. Hmm Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P What problems are you having ? We are using CentOS 4.2 Linux 2.6.9-22.0.1.ELsmp Asterisk SVN-trunk-r7353M I know, time to update, but I am not in the office currently and do *not* want to do it remotely ;) Julian Asterisk wrote: Hello Julian! As we are using the same HW (2850 and Sangoma cards) but have some problems with AMI (* manager interface) I wonder which OS (version of Linux, kernel version) are you using? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P We have a 2850 running with a Sangoma A102 , 50 agents using Agentcallbacklogin and around 4000 calls per day. No problems at all. With a te410 / 405 (we've got both, can't remember which one was in the dell) we had lockups almost every day. Julian. Klaus Darilion wrote: Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm2400p: fax detection not working
Giorgio Incantalupo wrote: I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I tried with a TDM400P and it worked at 80% (20% of faxes were lost). My test conf.files are: This is a known problem with the hardware echo canceler. For the time being, load the wctdm24xxp module with the 'vpmdtmfsupport=0' parameter so that the software tone detector will be used. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
This is an old version of trunk though - be careful :) Julian Asterisk wrote: We have problems with 'loosing' parts of messages sent from Asterisk Manager to our CTI Server (we also tested it with another test program, but the problem persisted). Sometimes (once per 4000 lines or so - depending on speed of network) manager improperly returns events. For example, one QueueMember will get overwritten by (or as part of) another, like this (see third line): Event: QueueMember Queue: 09 LocatiEvent: QueueMember Queue: 09 Location: Agent/09003 Membership: static Penalty: 1 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 We are using Red Hat ES Linux 4 (the same 2.6.9-22 kernel) but SVN-branch-1.2-r27093M. We also noticed that problem seems to be fixed in trunk, but we are not 'brave' enough to use trunk in production environment. But obviously your experiences with using trunk in production are quite good. Hmm Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P What problems are you having ? We are using CentOS 4.2 Linux 2.6.9-22.0.1.ELsmp Asterisk SVN-trunk-r7353M I know, time to update, but I am not in the office currently and do *not* want to do it remotely ;) Julian Asterisk wrote: Hello Julian! As we are using the same HW (2850 and Sangoma cards) but have some problems with AMI (* manager interface) I wonder which OS (version of Linux, kernel version) are you using? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, May 17, 2006 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P We have a 2850 running with a Sangoma A102 , 50 agents using Agentcallbacklogin and around 4000 calls per day. No problems at all. With a te410 / 405 (we've got both, can't remember which one was in the dell) we had lockups almost every day. Julian. Klaus Darilion wrote: Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I can not talk about Digium, but 2850 with Sangoma A102 works fine here (standard Debian 2.4.27 kernel) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID retain on internal transfer
Steve Davies wrote: In the cases previously mentioned, the user is doing an attended transfer using the handset features, and not Asterisk. I do not know whether SIP even allows the Caller ID to be changed at the point when two separate calls are bridged to one... It does, but Asterisk does not currently support that behavior (even in the development branch). I believe Olle's SIP transfer re-write may provide this functionality when Asterisk 1.4 is released, but I am not positive. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plan to free myself from AAH
Mimmus wrote: Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to rewrite dialplan from scratch and I'd like only to clean actual AAH dialplan. I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? Thanks It also works fine with the free CentOS version 3.? and 4.? Burn yourself a set of install CD's from the on line ISO's, do an install everything then install asterisk. You may want to compare and edit, rather than overwrite the conf files, there have been a bunch of changes, depending on how old the AAH version. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reading queue_logs
Hi, Are there any good free win32 apps for reading queue_logs? Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US. I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NO ringing tone while dialing
Hi! I have tried when the option 'm' , but I don?t want the default music on hold to be listened neither. I want nothing (silence) to be heard instead of ring, ring. Any idea how to do this??? The answer is in your question: Create a MOH music file with... silence in it. Then make a new MOH class for that and you are done. :-) Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I dont think the ping times are the issue. Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX crackilng
I apologize about doubling these up, I forgot the subject! I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I dont think the ping times are the issue. Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using REGEX function
Wes Santee wrote:
The first problem is obviously that the curly braces used in regex patterns
to
denote repeating patterns means something different to Asterisk. I would
expect
back-slashing to fix this. So...
This was just recently fixed in SVN branch 1.2, and the fix will be part
of the 1.2.8 release that will appear later this week.
In the meantime, you can work around the problem by storing the regex
string itself in a variable, then using a variable substitution in the
${REGEX()} function call, like this:
exten = ...,1,Set(MATCH=[2-9][0-9]{2}[2-9][0-9]{6})
exten = ...,n,Set(isnum=${REGEX(${MATCH} ${EXTEN:2})})
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX crackilng
I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this… Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I don’t think the ping times are the issue. If I had to troubleshoot the issue, I'd start with using Ethereal to see what type of traffic was happening every 30 seconds or so (causing the erratic ping times). Then I'd also be looking at 'iax2 show netstats' to see what the jitter, delay, and lost packets look like. If the ethereal packet trace indicates unusual traffic or broadcast storms, see if those can be corrected. Might even try 'fair weighted queuing' in the Cisco box to see what impact it may have (attempts to provide equal amounts of bandwidth to each session crossing the vpn). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Min-Expires
Samuel Tardieu wrote: Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the default_expirey globally. Yes, it should. Please open a bug report on bugs.digium.com with a 'sip debug' trace of this interaction so we can get it corrected. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diverse servers
I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system such that in event of failure, childAsterisk boxes, phones, ATAs, etc. can register to either box. I can handle the child's configuration, but how do I have it setup on the Asterisk boxes? I'm not exactly sure I explained this right, but hopefully someone can get what I'm talking about and ask further questions of me. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
Hi! We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I rebooted it a week ago. Yesterday the machine suddenly stop working, with a kernel panic. We was watching logs, and found in /var/log/asterisk just before the machine hung the messages posted avobe(is the first time we see it). Anyone know the cause? May 15 20:23:29 WARNING[4033]: PRI: !! Got reject for frame 65, but we only have others! May 15 20:27:49 WARNING[4033]: PRI: !! Got reject for frame 106, but we only have others! May 15 20:30:13 WARNING[4033]: PRI: !! Got reject for frame 114, but we only have others! May 15 20:36:10 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:37:48 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:38:37 WARNING[4033]: Failed to write frame May 15 20:39:08 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:39:14 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:39:27 WARNING[4033]: Avoided initial deadlock for 'Zap/65-1', 10 retries! May 15 20:39:49 WARNING[4033]: Avoided deadlock for 'Zap/65-1', 10 retries! May 15 20:39:57 WARNING[4033]: Avoided deadlock for 'Zap/54-1', 10 retries! May 15 20:39:57 WARNING[4033]: Avoided deadlock for 'Zap/65-1', 10 retries! May 15 20:39:57 WARNING[4033]: Prodding channel 'Zap/54-1' failed May 15 20:39:59 NOTICE[4033]: Unable to call channel Zap/g2/933238980 May 15 20:40:00 WARNING[4033]: Avoided deadlock for 'Zap/65-1', 10 retries! May 15 20:40:00 WARNING[4033]: Can't change device '**Unknown**' with no technology! May 15 20:40:07 WARNING[4033]: Avoided initial deadlock for 'Zap/66-1', 10 retries! May 15 20:40:42 WARNING[4033]: Avoided initial deadlock for 'Zap/65-1', 10 retries! May 15 20:41:13 WARNING[4033]: Avoided initial deadlock for 'Zap/65-1', 10 retries! May 15 20:41:25 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:41:34 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:41:34 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:41:34 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:41:34 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:41:35 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:42:07 WARNING[4033]: Avoided initial deadlock for 'Zap/66-1', 10 retries! May 15 20:42:34 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:42:56 WARNING[4033]: Avoided deadlock for 'Zap/66-1', 10 retries! May 15 20:43:05 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:43:22 WARNING[4033]: Avoided deadlock for 'Zap/7-1', 10 retries! May 15 20:43:22 WARNING[4033]: Avoided deadlock for 'Zap/7-1', 10 retries! May 15 20:43:35 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:43:53 WARNING[4033]: Avoided initial deadlock for 'Zap/69-1', 10 retries! May 15 20:43:56 WARNING[4033]: Avoided deadlock for 'Zap/67-1', 10 retries! May 15 20:45:04 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:45:21 WARNING[4033]: PRI: !! Got reject for frame 79, but we only have others! May 15 20:45:22 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:45:43 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:46:06 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:24 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:25 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:26 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:26 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:26 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:26 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:26 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:46:27 WARNING[4033]: Avoided deadlock for 'Zap/24-1', 10 retries! May 15 20:46:27 WARNING[4033]: Avoided deadlock for 'Zap/24-1', 10 retries! May 15 20:46:30 WARNING[4033]: Avoided deadlock for 'Zap/69-1', 10 retries! May 15 20:46:41 WARNING[4033]: Avoided deadlock for 'Zap/69-1', 10 retries! May 15 20:46:47 WARNING[4033]: Avoided deadlock for 'Zap/69-1', 10 retries! May 15 20:46:51 WARNING[4033]: Avoided initial deadlock for 'Zap/71-1', 10 retries! May 15 20:46:52 WARNING[4033]: Avoided deadlock for 'Zap/69-1', 10 retries! May 15 20:50:03 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:50:40 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10 retries! May 15 20:50:43 WARNING[4033]: Avoided initial deadlock for 'Zap/67-1', 10
Re: [Asterisk-Users] Sangoma A200D problem
Well, looks like we had a similar issue. Replaced the Sangoma and it worked. We have asked for a failure analysis from Sangoma on the defective card. Dr. Michael J. Chudobiak wrote: I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. Well, just to complete my own thread, this seems to be a probable hardware defect and Sangoma is sending a replacement. I had to live with software echo cancellation for a day or two - shudder - it's amazing how much better the hardware echo cancellation is! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
Ok, I going to do it, thanks El Martes, 16 de Mayo de 2006 17:06, Joshua Colp escribió: Edu wrote: Hi! We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I rebooted it a week ago. Yesterday the machine suddenly stop working, with a kernel panic. We was watching logs, and found in /var/log/asterisk just before the machine hung the messages posted avobe(is the first time we see it). Please contact Digium support about this issue. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Diverse servers
For your configuration to be like this RRDNS and Realtime. I believe someone made a patch for realtime to work correctly with RRDNS you would have to check the wiki or mantis to find it. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, May 17, 2006 9:51 AM To: Asterisk-users@lists.digium.com Subject: [Asterisk-Users] Diverse servers I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system such that in event of failure, childAsterisk boxes, phones, ATAs, etc. can register to either box. I can handle the child's configuration, but how do I have it setup on the Asterisk boxes? I'm not exactly sure I explained this right, but hopefully someone can get what I'm talking about and ask further questions of me. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi VoIP Handsets..
Andrew - I am not sure of the exact spectrum frequency which constitutes DECT in the US, but several major manufacturers are developing and selling DECT devices for the US market. Plantronics recently release a DECT 6.0 wireless headset, the CS55, which you can see here http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880 043/prod5430004 Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: Andrew Latham [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 9:08 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US. I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months.Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: CoryThe 480i-CT does not state DECT to my knowlege as the EU DECT standarduses reseved frequency space in the US.I have heard rumblings abouta US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity.AndrewOn 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!Hind sight is most always 20/20 or better.---___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi VoIP Handsets..
Not sure about overseas distribution for the Aastra 480i-CT, I am not aware of any VARs or distys across the pond. The UIP1868 has actually had production discontinued at Uniden. They are continuing to manufacture a Vonage Locked version of the unit, but have stopped producing the unlocked units. No word on a replacement as of yet. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: Colin MacMillan [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi VoIP Handsets..
I think around Q3/Q4 of this year, you'll see some very interesting new products which incorporate DECT for wireless. For consumer products with limited mobility, it seems to make a bit more sense than WIFI. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: Andrew Latham [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. Thanks Cory, DECT is very interesting and I hope that more devices come out soon... On 5/17/06, Cory Andrews [EMAIL PROTECTED] wrote: Andrew - I am not sure of the exact spectrum frequency which constitutes DECT in the US, but several major manufacturers are developing and selling DECT devices for the US market. Plantronics recently release a DECT 6.0 wireless headset, the CS55, which you can see here http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880 043/prod5430004 Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: Andrew Latham [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 9:08 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US. I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi VoIP Handsets..
According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months.Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: CoryThe 480i-CT does not state DECT to my knowlege as the EU DECT standarduses reseved frequency space in the US.I have heard rumblings abouta US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity.AndrewOn 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)[EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!Hind sight is most always 20/20 or better.---___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soekris hadware
Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a bigger box is needed? Any suggestions about where to pick up another box? 2) does the Digium TDM100P (already discontinued) fits fine in a soekris box? 3) running asterisk in a soekris 4801 SBC, what is the perfomance related to sip connections, analogue call quality and both mixed at the same time? 4) what is the actual state of fax support into asterisk / [EMAIL PROTECTED] ? 5) It's posible to create personalized dialplans that enables a hidden or passcode/password protected menu for remote administration or remote use of the pbx? Thanks in advance for your kind help and support. Jonathan GF -- si secretum tibi sit, tege illud, vel revela ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP debugging
Hi! I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. There is no log message, why this packet is not accepted/processed. Is there a ways to increase the sip debugging? thanks klaus Retransmitting #5 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' Retransmitting #6 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overwriting SIP headers
I'm wondering if anyone has a solution to this before I begin looking at making some changes to the SIP channel. Basically when calling SIPAddHeader() twice from the Dialplan or an AGI script with the same header name it adds duplicate headers instead of overwriting the existing one. Here's a practical situation where this applies. A call is to be terminated via SIP and we have two possible routes to terminate the call and the first requires a static value in the P-Asserted-Identity header and the second requires a variable value in the same header. We do all of our setup and then call SIPAddHeader to add our P-Asserted-Identity with the static value and call Dial(). An error occurs so we wish to fall back to the second provider, more setup work is done and SIPAddHeader is called to add the new value the second provider wants but when Dial is called this time instead of only the new value being sent both values are sent and the request contains two P-Asserted-Identity headers which causes problems with the second provider. So, does anyone have a creative solution to this? I've already tried to loop through the SIPADDHEADER0X variables and unset the previous values (well set them to ) but it didn't help and unfortunately AGI doesn't provide a way to delete variables. Thanks, -Evan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax asterisk 1.2
Hi, I have a problem with fax in asterisk version 1.2.7. The application rxfax doesn't work triggering these errors : channel.c: Dropping incompatible voice frame on ... of format slin since our native format has changed to ulaw chan_sip.c: sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) I think it's a problem of transcoding from ulaw to slin but I can't see how to fix it : which file .c i must modify or !! Can anyone please tell how to solve this problem?!! Thank u very much in advance _ Windows Live Mail : découvrez et testez la version bêta ! http://www.ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP redirect
Hi, is it possible to let asterisk issue a SIP redirect? A SIP invite command by a SIP client should be answered by 30X Temporarly moved to SIP/ Is this possible with asterisk, maybe from within the dialplan? (reinvite is not what I'm looking for, because it does not completely release the originally called SIP server, e.g. if reinvite fails, ...) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP debugging
Klaus Darilion wrote: I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. SVN trunk is not Asterisk 1.2. There is no way to help you with this partial SIP trace, and without any Asterisk version or configuration information. Asking 'smart questions' usually leads to people being able to help you :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP redirect
Roger Schreiter wrote: is it possible to let asterisk issue a SIP redirect? A SIP invite command by a SIP client should be answered by 30X Temporarly moved to SIP/ Have you read any documentation on the applications available in Asterisk, or on the voip-info wiki? The Transfer() dialplan application will do exactly this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Diverse servers
was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060517/d9af5983/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a WRT54G?
What do you mean exactly? Do you want to flash Asterisk on this Wireless Router? If so, I would like to hear more such as what kind of end-user applications would you like to run on it? Cheers Fifou Frank Tarczynski a écrit : I'm looking for a recent asterisk package for the Linksys WRT54G. Has anyone know of a 1.2.X build for this box? Thanks, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soekris hadware
Google and voip-info.org will have answers to all of your questions.On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote:Hi group,i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here.My question are the following:[...] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soekris hadware
more kindly : http://www.astlinux.org/ Olivier Christopher Snell a crit: Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP Gateway behind NATS - SIP/2.0 404 Not Found
Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My sip.conf looks somewhat like this [general] context=default; Default context for incoming calls bindport=5065; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls externip = 83.xxx.xxx.xxx; Address that we're going to put in outbound SIP messages localnet=10.2.70.0/255.255.255.0 localnet=192.168.18.0/255.255.255.0; All RFC 1918 addresses are local networks [thephone] type=peer host=thephonedomain.com port=5065 username=abcd nat=no usereqphone = yes ;canreinvite=no If I made a call to local SIP phone, it works fine. But to the SIP phone outside the NAT, it just doesn't seem to work. I have no idea what else I should do. Anybody could give me some suggestion?? regards, Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a WRT54G?
Check out http://www.openwrt.org I run their prebuilt asterisk-1.0.7 package on a Linksys WRT54GL and it works like a charm, just don't expect too much performance out of it :) -Evan Phi Fou wrote: What do you mean exactly? Do you want to flash Asterisk on this Wireless Router? If so, I would like to hear more such as what kind of end-user applications would you like to run on it? Cheers Fifou Frank Tarczynski a écrit : I'm looking for a recent asterisk package for the Linksys WRT54G. Has anyone know of a 1.2.X build for this box? Thanks, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plan to free myself from AAH
On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it. The problem I have with AAH / AMP / FreePBX is that the configuration files are absolutely full of useless garbage and are really not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configure asterisk manually. Rebuild everything one step at a time so that the functionality remains as you'd like it to be, but that the actual configs aren't full of that FreePBX garbage :) -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a WRT54G?
It also looks like they've got a 1.2 package for openwrt now: https://dev.openwrt.org/browser/trunk/openwrt/package/asterisk/Makefile -Evan Evan Borgström wrote: Check out http://www.openwrt.org I run their prebuilt asterisk-1.0.7 package on a Linksys WRT54GL and it works like a charm, just don't expect too much performance out of it :) -Evan Phi Fou wrote: What do you mean exactly? Do you want to flash Asterisk on this Wireless Router? If so, I would like to hear more such as what kind of end-user applications would you like to run on it? Cheers Fifou Frank Tarczynski a écrit : I'm looking for a recent asterisk package for the Linksys WRT54G. Has anyone know of a 1.2.X build for this box? Thanks, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plan to free myself from AAH
I wouldn't knock the third party friendly interfaces to Asterisk too hard. They will evolve and improve over time. The adoption of Asterisk as a mainstream PBX is dependent upon a user friendly interface. On 5/17/06, Strom Carlson [EMAIL PROTECTED] wrote: On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it.The problem I have with AAH / AMP / FreePBX is that theconfiguration files are absolutely full of useless garbage and arereally not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configureasterisk manually.Rebuild everything one step at a time so that thefunctionality remains as you'd like it to be, but that the actualconfigs aren't full of that FreePBX garbage :) --Strom Carlsonhttp://www.stromcarlson.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Plan to free myself from AAH
What I did with AMP was take the best parts of it and copy/paste to a clean extensions.conf, then add my modifications onto it. Worked for me. -Original Message- From: Strom Carlson [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Plan to free myself from AAH On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it. The problem I have with AAH / AMP / FreePBX is that the configuration files are absolutely full of useless garbage and are really not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configure asterisk manually. Rebuild everything one step at a time so that the functionality remains as you'd like it to be, but that the actual configs aren't full of that FreePBX garbage :) -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Listening on Multiple Interfaces
Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever on multiple network interfaces? Thanks, Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using REGEX function
Kevin P. Fleming wrote:
Wes Santee wrote:
This was just recently fixed in SVN branch 1.2, and the fix will be part
of the 1.2.8 release that will appear later this week.
In the meantime, you can work around the problem by storing the regex
string itself in a variable, then using a variable substitution in the
${REGEX()} function call, like this:
exten = ...,1,Set(MATCH=[2-9][0-9]{2}[2-9][0-9]{6})
exten = ...,n,Set(isnum=${REGEX(${MATCH} ${EXTEN:2})})
Thanks! This looks like a good work-around until the new release is out.
Cheers,
-Wes
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can two asterisk servers share the same dialplan by using FreePBX?
Hi All: Assume that FreePBX can configure a remote MySQL server, is any way that I could use FreePBX to configure second Asterisk by using the same dialplan deployed for configuring the first Asterisk server? If answer is no, could anybody help me some ideas to resolve this? Thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asus P5GD1... anyone using with Asterisk ??
Hello, anyone is using the Asus P5GD1-Pro with Asterisk ? For a customer I need to use this mainboard for 2 PCI Boards (1 BNS8 Beronet 8xRDSI and 1 TDM2400 ). Any reference for bad or good mainboard for asterisk ? This is a Intel 915P Chipset. Greetings to all. Juan Carlos Valero. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
just have to say WOW I got a new voip wifi handset. Not yet on the market, but the constructor promised me international versions if I can have a 50+ order. Well, the constructor is a very well known handset provider in the Pstn/Isdn world and Skype world :( ... It's a german one... Smell good, go on, it's S.s, they will put the handset on the market around september. Honestly, I gave a few calls with it, dunno about battery life and so on, quality is very very good, but what I can say is that the specs are WONDERFULL. Linux based :) Olivier ps: public price will be around 199 taxes includes. If I have 50+ orders, I promise to do my best to have the best price for all, this is NOT a commercial offer, just an offer for asterisk users (also ser users). Kind of open source hardware offer ;) If any of you can host the specs, I will send a pdf The VoIP Connection a crit: According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP debugging
Kevin P. Fleming wrote: Klaus Darilion wrote: I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. SVN trunk is not Asterisk 1.2. Of course - sorry. I've meant Asterisk 1.2 from SVN branch 1.2 There is no way to help you with this partial SIP trace, and without any Asterisk version or configuration information. Asking 'smart questions' usually leads to people being able to help you :-) IMO this was a smart question. I did not asked to debug my call flows, but I asked how can I debug it myself. For some reason Asterisk does not like my SIP responses, but there is no Warning, Error or any other log message although verbose=9 and sip debug. Shouldn't there be some error indication if Asterisk discards a response? thanks Klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Using Multiple Databases with ODBC?
Hello! Does anyone know how I can use two diferent databases with ODBC with the same erealtime family? Something like this: res_odbc.conf: ;;; odbc setup file [ast_cnf1] dsn = ORACLE username = asterisk password = asterisk pre-connect = yes [ast_cnf2] dsn = MySQL username = asterisk password = asterisk pre-connect = yes and the extconfig.conf: voicemail = odbc,ast_cnf1,voicemail_conf voicemail = odbc,ast_cnf2,voicemail_conf Thanks, jsalas. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird Error Upgrading 7960's to 8.2SIP
Hello Everyone, I just picked up 10 new 7960G. Im trying to load 8.2 on them but getting a weird error during the tftp transaction when reading OS79XX.TXT Here is what im getting. Ben Blakely, CISSP Security Architect Blink Communications div. of Oakville Hydro 905-825-6369 905-466-1086 sip: [EMAIL PROTECTED] Fingerprint: 716A E001 FAE6 DE93 4CBF BC92 5C9B 5B48 3925 1585 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Listening on Multiple Interfaces
I have IAX on a server with dual nic in 2 subnets, I can connect to the IAX channel on both subnets, I do not have a bind setting in my IAX.conf file. I don't know if that will work for SIP/RTP, but I would think so. On 5/17/06, Douglas Garstang [EMAIL PROTECTED] wrote: Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever on multiple network interfaces?Thanks,Doug___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird Error When upgrading 7960G to 8.2
Hello All, I just picked up 10 new 7960Gs and having a hardtime upgrading the firmware on them. We already have 30 or 40 of these phones in production. Typically when I get a new phone, I just plug it into the voice vlan and it auto grades to the firmware in OS79XX.txt. Now whats happening is im getting a weird error during the reading of that file. Heres the output from the TFTP logs. May 17 16:49:31 asterisk in.tftpd[31683]: sending NAK (1, File not found) to 192.168.10.38 May 17 16:49:40 asterisk in.tftpd[31684]: RRQ from 192.168.10.34 filename OS79XX.TXT May 17 16:49:40 asterisk in.tftpd[31684]: sending NAK (4, Request not null-terminated) to 192.168.10.34 May 17 16:49:40 asterisk in.tftpd[31685]: RRQ from 192.168.10.34 filename SEP001647051680.cnf.xml May 17 16:49:40 asterisk in.tftpd[31685]: sending NAK (1, File not found) to 192.168.10.34 Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Providers using Embedded Devices
Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asus P5GD1... anyone using with Asterisk ??
Just turned up a system with an A8N/Athlon 64 on Monday, 40 extensions, 6 IAX locations w/ TE-110P and TDM400. Works perfect, no echo, ZTTEST runs steady at 99.9873%. Had to disable everything, onboard LAN, RAID, USB, serial, etc. Replaced the onboard LAN with an Intel 82557 based card, because the A8N ships with some goofy Yukon chipset and the driver sucks. This was contrary to my gut telling me to never, ever do a production system with an Athlon, but so far so good. -Original Message- From: Información Capa Tres S.L. [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asus P5GD1... anyone using with Asterisk ?? Hello, anyone is using the Asus P5GD1-Pro with Asterisk ? For a customer I need to use this mainboard for 2 PCI Boards (1 BNS8 Beronet 8xRDSI and 1 TDM2400 ). Any reference for bad or good mainboard for asterisk ? This is a Intel 915P Chipset. Greetings to all. Juan Carlos Valero. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plan to free myself from AAH
On 5/17/06, David K Parker [EMAIL PROTECTED] wrote: I wouldn't knock the third party friendly interfaces to Asterisk too hard. They will evolve and improve over time. The adoption of Asterisk as a mainstream PBX is dependent upon a user friendly interface. Well, as soon as a GUI shows up that doesn't make configuring Asterisk like trying to sew with boxing gloves on, I'll give it a good, hard, unbiased look. For now, though, the available interfaces are really just not there yet - they don't allow enough flexibility and they are very easy to outgrow. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Listening on Multiple Interfaces
Douglas Garstang wrote: Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever on multiple network interfaces? Yes, I have a system setup to listen to both a 10.10.10.x network and 192.168.102.0 network. I don't specify a bind address. The phone on site are on the 10.10.10.x network, the phone I carry around with me it located on the 192.168.x.x network. It seems to work well. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
Hello: I have tested the Zyxel wifi phone with locustworld AP's. The roaming between the AP's is seemless but works. MohamoudOn 5/17/06, olivier.taylor [EMAIL PROTECTED] wrote: just have to say WOW I got a new voip wifi handset. Not yet on the market, but the constructor promised me international versions if I can have a 50+ order. Well, the constructor is a very well known handset provider in the Pstn/Isdn world and Skype world :( ... It's a german one... Smell good, go on, it's S.s, they will put the handset on the market around september. Honestly, I gave a few calls with it, dunno about battery life and so on, quality is very very good, but what I can say is that the specs are WONDERFULL. Linux based :) Olivier ps: public price will be around 199€ taxes includes. If I have 50+ orders, I promise to do my best to have the best price for all, this is NOT a commercial offer, just an offer for asterisk users (also ser users). Kind of open source hardware offer ;) If any of you can host the specs, I will send a pdf The VoIP Connection a écrit: According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
[Asterisk-Users] Re: DISA SPA3000 issues
I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such feature. Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
