Re: [Asterisk-Users] Do I *need* to compile zaptel?
You'll need to get the kernel source for 2.6.7. apt-get install kernel-source.2.6.7 John, thanks for that. Which version of Asterisk are you using? head or stable? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using Voipjet?
On Sat, 30 Oct 2004 11:12:39 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: With so long distances, there is nothing better than G.729. And why would G.729 be any better than iLBC or Speex. In fact, if you have packet loss, G.729 absolutely sucks. iLBC is far more forgiving and sound quality is just as good. The only reason to use G.729 at all is installed base in handsets. Once iLBC will be supported by the major DSP chipsets (ie Texas Instruments as of next year) and more handsets will have support for iLBC, the reason to use G.729 will vanish. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g723 in pass-thru mode asterisk
Hi, I am trying to dial out from asterisks to a h323 endpoint. But it's not allowing to use g723. How can I get it to work in pass-thru mode. Current scenerio is H323 Ep1===Asterisk===Dial-Out=H323Ep2 This works with 711 but not with 723. g723 is there on both the Ep. As H323 Ep1==Gatelkeeper=H323Ep2 works filne with h323. I am using the chan_h323 driver provided with Asterisk. How can I get the g723 to work in pass thru mode. Regards, Prashant Ownmail LLC. 2 is not equal to 3 -- not even for large values of 2. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wireless phones connected to VOIP DECT base station
Hi list! I found an interesting wireless phone product. Tiptel will be selling a base station for DECT phones that is VOIP capable. The base station comes in two models, one with the SCCP the other with H.323 protocol support. The interesting bit is that the base station is VOIP connected but you can use standard (cheap) DECT phones to connect to it. I found it on www.tiptel.nl (the websites in other countries do not mention this model) and the model is called tiptel DECT-Z 600 IP systeem Yu can find it on www.tiptel.nl - producten - DECT Draadloze telefoonsystemen - tiptel DECT-Z 600 IP systeem http://www.tiptel.nl/fr_top.asp?lang_id=1mid=901pid=901 Unfortunately the product page is in Dutch only. I wonder if Tiptel make this themselves or whether they bough an OEM product from a manufacturer. When googling I could not find a similar product though and there aren't nay mentioned under Wireless VOIP in the wiki either. Are there other similar base stations? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests. I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. Where is the problem Can anyone help me? here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; 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 disallow=all allow=alaw allow=ulaw allow=gsm allow=g729 [xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid=Savas Pavlidis 1239 host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT [10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729 [419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729 EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten = _3XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _3XX,n,Congestion ; ; ; exten = 419,1,Dial(SIP/419) exten = 420,1,Dial(SIP/xlite1) exten = 420,2,Congestion ; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc. CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 ! begin:vcard fn:Savas Pavlidis n:Pavlidis;Savas email;internet:[EMAIL PROTECTED] tel;work:+30 2310 573300 tel;fax:+30 2310 752280 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO flash from sip phone
Hi All! I am trying to flash a fxo line from my sip phone during a call, in order to hold or transfer the call. Could someone please tell me how to do this, if it is possible. Everything in zapata.conf about transfer is enabled, and my incoming POTS line support this (I have checked with a analog headset) Thanks and rgds Gunnar Anderson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
Hi Remco, I found it on www.tiptel.nl (the websites in other countries do not mention this model) and the model is called tiptel DECT-Z 600 IP systeem Yu can find it on www.tiptel.nl - producten - DECT Draadloze telefoonsystemen - tiptel DECT-Z 600 IP systeem http://www.tiptel.nl/fr_top.asp?lang_id=1mid=901pid=901 Unfortunately the product page is in Dutch only. I wonder if Tiptel make this themselves or whether they bough an OEM product from a manufacturer. When googling I could not find a similar product though and there aren't nay mentioned under Wireless VOIP in the wiki either. It seems to be made by KIRK. Here is a link I found: http://www.kirktelecom.com/company/suk110.asp No pricing found so far. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * and Verisign SIP-7 service
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP mediation ... what does it not work? I don't know the particulars, because I've never used (or even looked at MGCP). All I know is that whenever the issue comes up, people here say that Asterisk does not know how to act as an MGCP Gatekeeper, only as an agent. I presume it would have to act as a gatekeeper to control an MGCP-based media gateway, because those devices are all intended to be controlled by some sort of softswitch. IMO, there is no such thing as an MGCP gatekeeper; try that phrase with Google and it will be obvious. Gatekeeper is an H.323 term. MCGP is a master-slave protocol. The master is referred to as a Call Agent, a Media Gateway Controller, or just a softswitch. This is the role that Asterisk can play. The slave is a Media Gateway, an MGCP phone, an MGCP ATA, or just an endpoint. Asterisk cannot presently act as a slave. Of course, any large system may have higher-level elements that handle authorization, accounting, complex routing, queueing, etc., but those topics are beyond the scope of MGCP. Perhaps the term gatekeeper was used in that context. So, I think that Asterisk will provide the functionality that you desire. However, I don't know if SIP-MGCP calls can presently be completed without Asterisk proxying the media stream, so you may have performance issues. Perhaps someone else can address that. --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swissvoice IP10S opinions?
Hi, On Sat, 2004-10-30 at 02:50, JB Hewit wrote: Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? Use the MGCP firmware, its a lot more mature than their new SIP version. I'm currently rolling out several hundreds of them and they make excellent 'standard issue' desk phones. Asterisk could get a bit more work in the Business package support, but thats all fancy. There was an issue with RTP, because the IP10 can only deal with 1 RTP stream at a time, which is why I asked Digium to implement the singlepath option :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggestion re: SIP/NAT/*
Michael Giagnocavo wrote: I think his point is that for a commercial rollout (say, a VSP), IAX is not practical for all clients right now. It's not strange to have a personal preference that is technically better but not commercially viable. That's not an insult, just how things are sometimes. Maybe if there were some ~$70 NAT router/gateway/bridge/UPnP/etc./etc. devices that supported IAX, this'd change. Sorry what are you wanting the NAT router/gateway/bridge/UPnP/etc./etc. devices to support about IAX exactly? It does not require any mad packet mangling like SIP does. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggestion re: SIP/NAT/*
Exactly. - Original Message - From: Michael Giagnocavo [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, October 29, 2004 10:56 PM Subject: RE: [Asterisk-Users] Suggestion re: SIP/NAT/* I think his point is that for a commercial rollout (say, a VSP), IAX is not practical for all clients right now. It's not strange to have a personal preference that is technically better but not commercially viable. That's not an insult, just how things are sometimes. Maybe if there were some ~$70 NAT router/gateway/bridge/UPnP/etc./etc. devices that supported IAX, this'd change. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, October 29, 2004 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/* On Fri, 2004-10-29 at 21:53 -0400, Steve Totaro wrote: Probably since there are so many SIP devices out there now and only a couple IAX. In the future it is an awsome replacement. So you would rather drive a '70s pinto instead of a Bugatti because there are more 70's fire bomb pintos? - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 7:57 PM Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/* --SNIP ALL-- IAX is no adequate replacement option for SIP either. --SNIP ALL-- What?! How on earth could you come to that conclusion?! -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggestion re: SIP/NAT/*
- Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, October 30, 2004 6:18 AM Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/* Michael Giagnocavo wrote: I think his point is that for a commercial rollout (say, a VSP), IAX is not practical for all clients right now. It's not strange to have a personal preference that is technically better but not commercially viable. That's not an insult, just how things are sometimes. Maybe if there were some ~$70 NAT router/gateway/bridge/UPnP/etc./etc. devices that supported IAX, this'd change. Sorry what are you wanting the NAT router/gateway/bridge/UPnP/etc./etc. devices to support about IAX exactly? It does not require any mad packet mangling like SIP does. -- Cheers, Matt Riddell I think he meant something along the lines of what some people are trying to do with the Linksys wrt54g. Have the router not only forward the packets but actually speak the language and be able translate for internal SIP clients. A mini asterisk box. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is VERY interesting -- A gateway betweenproprietary digital sets and SIP?
I have delt with their 3com offerings and yes if you are lucky enough to be able to use this as a stepping stone solution then its a closed deal (on 3com system) - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 30, 2004 1:00 AM Subject: [Asterisk-Users] This is VERY interesting -- A gateway betweenproprietary digital sets and SIP? Has anyone had any experience with these folks? http://www.citel.com/index/index.asp That could be a compelling way to displace a legacy system with an Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA() anyone?
On Fri, Oct 29, 2004 at 09:50:47PM -0400, Nick Bachmann wrote:
Michael George wrote:
I'm having some trouble with DISA() in a call plan that worked before 1.0.
If
anyone has experience with it, I would appreciate some advice.
Perhaps you could post relavent sections of your dialplan...?
Thanks for your reply!
Here's part of my dialplan on my home machine. It's running
CVS-HEAD-09/21/04, and my zap and sip phones have this initial context. I
can, from either zap or sip, hit 8 and get a new dialtone and then I can enter
6 and get the audio from tt-allbusy.
[internal]
; here for timeout, invalid, and park expire
;exten = s,1,Background(invalid)
exten = s,1,NoOp(start of internal)
;exten = s,2,Playtones(dial)
exten = s,2,DISA(no-password,internal)
exten = 6,1,Playback(tt-allbusy)
exten = 8,1,goto(s,1)
On my machine at work, running CVS-v1-0-10/28/04, I have the following
section of dialplan. My sip phones have this as their initial context and if
I dial from home (through IAX), my dialplan sends me to internal,s,1, also.
Whenever I get to DISA(), either through IAX2 from another * box or by
pressing 6 on the sip phone, I will get another dialtone from DISA(), as I
would expect. However, no matter what key I press next, I get a hangup from
this system.
I have tried notransfer=yes and notransfer=no and that doesn't seem to make a
difference. When I have a chance, I plan to try upgrading my home system to *
v1.0.x and see if it will then also fail, but I'm hoping there is something
that one more experienced with * and DISA() can point out first.
[internal]
ignorepat = 9
include = parkedcalls ; allow parking
exten = s,1,NoOp(${CONTEXT}, ${EXTEN}, ${PRIORITY})
;exten = s,2,Background(vm-enter-num-to-call)
exten = s,2,DISA(no-password, ${CONTEXT})
exten = 6,1,GoTo(s,1)
exten = 773,2,Playback(tt-allbusy)
Thanks for any help anyone might have.
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
___
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Ang: [Asterisk-Users] FXO flash from sip phone
An update on my own question...There is some built in numbers in the Zap channel. I think that *0 should do a hook flash but nothing happens. What have I missed? [EMAIL PROTECTED] 2004-10-30 10:50:32 Hi All! I am trying to flash a fxo line from my sip phone during a call, in order to hold or transfer the call. Could someone please tell me how to do this, if it is possible. Everything in zapata.conf about transfer is enabled, and my incoming POTS line support this (I have checked with a analog headset) Thanks and rgds Gunnar Anderson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This is VERY interesting -- A gatewaybetweenproprietary digital sets and SIP?
Thanks. I'll have to see about that SIP functionality. [EMAIL PROTECTED] wrote: I have delt with their 3com offerings and yes if you are lucky enough to be able to use this as a stepping stone solution then its a closed deal (on 3com system) - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 30, 2004 1:00 AM Subject: [Asterisk-Users] This is VERY interesting -- A gateway betweenproprietary digital sets and SIP? Has anyone had any experience with these folks? http://www.citel.com/index/index.asp That could be a compelling way to displace a legacy system with an Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic Card + TP100B
Hi; I have a Dialogic Card D160 SC and Trunk Board TP100B installed on my PC, I configured them and was able to detect the Trunck Board and the Dialogic Card, but I am not able to ping the TP100B? Is there any one has idea about special settings to be done for TP100B to be able to ping it? Regards Bilal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] loss concealment
Is asterisk capable of sealing (some amount) of losses that occur on IP based channels before it routes the Calls to a TDM channel (BRI, E1, etc.) to limit quality loss if IP loss occurs ? chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modifying CDR data?
Can someone please help me out here? On Oct 29, 2004, at 10:37 AM, Roy Sigurd Karlsbakk wrote: hi I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax registration port number
I'm trying to config a temp iax connection between two current * boxes. One is behind a firewall, the other uses a registered IP. I config'ed the * box behind the firewall to 'register' with the one that has a registered IP. The registration is occuring and the CLI indicates: -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50192 -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50262 However, I'm confused about the :50192 and :50262 port numbers shown above. I was expecting to see udp 4569 instead. I was hoping to use the registration process to avoid having to write a firewall rule allowing udp 4569 inbound. Am I off base or just missing something simple here? How do I force registration to use udp 4569 for both source and destination ports instead of the changing port numbers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On October 29, 2004 11:49 pm, Chris A. Icide wrote: Only in the X100P format, and only 2 of them I have to ask -- why are you running such high-end equipment for a craptastic FXO device? Don't you find other issues that going to a TDM4xxP or even a T1+channel bank would fix? I mean I ran an X100P and TDM410P (1FXO+1FXS) on a weenie P90 for cryin' out loud... What's this super big system getting you (except maybe future proofing?) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using Voipjet?
On October 30, 2004 02:38 am, Benjamin on Asterisk Mailing Lists wrote: And why would G.729 be any better than iLBC or Speex. lower conversion latency? less bandwidth? In fact, if you have packet loss, G.729 absolutely sucks. iLBC is far more forgiving and sound quality is just as good. Um... Packet Loss Concealment is *not* currently implemented in Asterisk. iLBC is currently no different than GSM or even ULAW on Asterisk in this respect. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax registration port number
Rich Adamson wrote: I'm trying to config a temp iax connection between two current * boxes. One is behind a firewall, the other uses a registered IP. I config'ed the * box behind the firewall to 'register' with the one that has a registered IP. The registration is occuring and the CLI indicates: -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50192 -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50262 However, I'm confused about the :50192 and :50262 port numbers shown above. I was expecting to see udp 4569 instead. I was hoping to use the registration process to avoid having to write a firewall rule allowing udp 4569 inbound. Am I off base or just missing something simple here? How do I force registration to use udp 4569 for both source and destination ports instead of the changing port numbers? The NAT router is translating the SOURECE port, which is perfectly fine. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP to SIP echo problem
I have a dual xeon server with 2 MB of ram running latest CVS * all calls are SIP and mu-law is the default codec for all connections my cpu power has not ever reach above 30% of its load. everything works fine. the problem is an echo on caller side of a call. but this is not an always event, it occurs random. what could cause echo on sip connections? please help thanks, Paradise Dove ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax registration port number
Rich Adamson wrote: I'm trying to config a temp iax connection between two current * boxes. One is behind a firewall, the other uses a registered IP. I config'ed the * box behind the firewall to 'register' with the one that has a registered IP. The registration is occuring and the CLI indicates: -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50192 -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50262 However, I'm confused about the :50192 and :50262 port numbers shown above. I was expecting to see udp 4569 instead. I was hoping to use the registration process to avoid having to write a firewall rule allowing udp 4569 inbound. Am I off base or just missing something simple here? How do I force registration to use udp 4569 for both source and destination ports instead of the changing port numbers? The NAT router is translating the SOURECE port, which is perfectly fine. Okay, I can buy that. But when I route an iax call from the registered IP * to the one hiding behind the firewall, the firewall gives an immediate icmp port unreachable as the call setup uses 4569/4569 ports (not the register ports). Maybe my Dial(IAX... is messed up then? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax registration port number
Rich Adamson wrote: Rich Adamson wrote: I'm trying to config a temp iax connection between two current * boxes. One is behind a firewall, the other uses a registered IP. I config'ed the * box behind the firewall to 'register' with the one that has a registered IP. The registration is occuring and the CLI indicates: -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50192 -- Registered 'xxx2yyy' (AUTHENTICATED) at 1.2.3.193:50262 However, I'm confused about the :50192 and :50262 port numbers shown above. I was expecting to see udp 4569 instead. I was hoping to use the registration process to avoid having to write a firewall rule allowing udp 4569 inbound. Am I off base or just missing something simple here? How do I force registration to use udp 4569 for both source and destination ports instead of the changing port numbers? The NAT router is translating the SOURECE port, which is perfectly fine. Okay, I can buy that. But when I route an iax call from the registered IP * to the one hiding behind the firewall, the firewall gives an immediate icmp port unreachable as the call setup uses 4569/4569 ports (not the register ports). Maybe my Dial(IAX... is messed up then? Are you using Dial(IAX2/[EMAIL PROTECTED]) or Dial(IAX2/iaxconfentry) If you are dialing by IP then Asterisk doesn't know anything about the existing connection and won't know the correct port information. If you ARE dialing by iax.conf entry I don't know what it will do, but if you are dialing by IP adddress than I KNOW it won't work without port forwarding on the NAT router. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
we use RHAS and whitebox and are quite happy with it on heavy loaded boxes. Dunno about analog stuff tho since we don't use it :) On Sat, 30 Oct 2004 09:57:27 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On October 29, 2004 11:49 pm, Chris A. Icide wrote: Only in the X100P format, and only 2 of them I have to ask -- why are you running such high-end equipment for a craptastic FXO device? Don't you find other issues that going to a TDM4xxP or even a T1+channel bank would fix? I mean I ran an X100P and TDM410P (1FXO+1FXS) on a weenie P90 for cryin' out loud... What's this super big system getting you (except maybe future proofing?) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On Sat, 30 Oct 2004 09:57:27 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On October 29, 2004 11:49 pm, Chris A. Icide wrote: Only in the X100P format, and only 2 of them I have to ask -- why are you running such high-end equipment for a craptastic FXO device? The X100P is not only an FXO device. Many folks use it as a Zaptel timing source only. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * and Verisign SIP-7 service
Stewart Nelson wrote: MCGP is a master-slave protocol. The master is referred to as a Call Agent, a Media Gateway Controller, or just a softswitch. This is the role that Asterisk can play. The slave is a Media Gateway, an MGCP phone, an MGCP ATA, or just an endpoint. Asterisk cannot presently act as a slave. Thanks for clearing that up... as I said in my message, I'd never even looked at MGCP. Now at least I know a little more about it :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modifying CDR data?
Roy, On Sat, 2004-10-30 at 15:31, Roy Sigurd Karlsbakk wrote: I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this? I don't think you can change dst from the extension flow just like that (maybe via an app, but that might have alternate consequences) I've done some scripting with entirely different purposes, but it may fit your needs: create an AGI script that is called when a call comes in, use that to store the uniqueid of the call leg into a database. Then check if call diversion is active and log that too. Afterwards, check (i.e. once an hour or whatever is convenient) and match CDR versus your own database. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On October 30, 2004 10:37 am, Benjamin on Asterisk Mailing Lists wrote: The X100P is not only an FXO device. Many folks use it as a Zaptel timing source only. Yes you are of course correct; but that then raises the question of why he wants two in there. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Config Files - searching
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick Sent: Friday, October 29, 2004 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP 500 Config Files - searching All, Has anyone got some example config files for the Polycom IP 500 SIP phones?Specifically the sip.cfg, ipmid.cfg phonename.cfg and any others that are needed to get the phones registered with *. Scott, I have config files that work well for my Polycom 300, 500, and 600. Uncompressed it's about 200K, too big to post here, but I will email them to you privately. If anyone else is interested, please email me off list The configs are very basic, not wavering too much from the Polycom defaults. For example, I haven't modified any SIP paramaters, voice quality, echo, etc. It's just for a basic phone with a single line appearance. Reid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using Voipjet?
On Fri, 29 Oct 2004 20:13:02 +0200, Wilson Pickett wrote: I've used them for calls terminating in the US with good results. I happened to put through a call to Romania today and it seemed the person was hearing me very much lagged behind. The actual asterisk IAX figure given was like 80 ms which is usually pretty decent for talking to someone. I've been using VoipJet since early August. I don't put a huge number of minutes through them, but I have been very impressed with the service. My dialplan tries them first, then falls back to trying NuFone and VoicePulse Connect if the VJ connection is down. It's almost never been down. They did for a short while have a problem passing DTMF but John got that fixed quickly. Also, their handling of CDR was not usefull at first, but hase been improved recently. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 !michael.tag ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
On Sat, 30 Oct 2004 09:37:00 +0200 (CEST), Remco Barende wrote: Hi list! I found an interesting wireless phone product. Tiptel will be selling a base station for DECT phones that is VOIP capable. The base station comes in two models, one with the SCCP the other with H.323 protocol support. The interesting bit is that the base station is VOIP connected but you can use standard (cheap) DECT phones to connect to it. I found it on www.tiptel.nl (the websites in other countries do not mention this model) and the model is called tiptel DECT-Z 600 IP systeem Yu can find it on www.tiptel.nl - producten - DECT Draadloze telefoonsystemen - tiptel DECT-Z 600 IP systeem http://www.tiptel.nl/fr_top.asp?lang_id=1mid=901pid=901 Unfortunately the product page is in Dutch only. I wonder if Tiptel make this themselves or whether they bough an OEM product from a manufacturer. When googling I could not find a similar product though and there aren't nay mentioned under Wireless VOIP in the wiki either. Are there other similar base stations? Please forgive my ignorance, but what exactly is DECT? I don't see the term referenced anywhere in North America. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 !michael.tag ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF and codec
hi the SIP doc on the wiki said that DTMF inband only worked on G.711. If I use G.726 and RFC2833 DTMF, will asterisk re-generate the DTMF at the Zap interface? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: asterisk SER and grandstream
hi list, anyone have any success getting asterisk to pass message waiting indicator to a grandstream with SER in the middle as a SIP proxy? I recently implemented SER between asterisk and my SIP clients and it's significantly more stable (no more dropped clients) but i haven't been able to figure out how to send message waiting indication so the grandstream's LCD flashes. If anyone has succesfully done this i'd be grateful for the info. thanks, yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latency/delay on IN1002 - PA1688 phone
Hello, I have just bought an IN1002 phone from Integrated Networks in Chine ( www.integratednetworks.com.cn). This phone is based on the 50 MHz PA1688 processor like many others and I find I have quitea lot of delay/latency/echo with the phone when using SIP. I have noticed this when directly connecting to my SIP provider (SIPPhone), which could be their network, but I also find the same thing when I am just using my mini-Asterisk system without going out to my SIP provider. I am not seeing the same degree of delay with my IAX softphone, so I am thinking that the problem is the IN1002. Does anyone have any ideas? The web configuration page for the phone has IAX as a protocol, but the folks at Integrated Networks tell me this has not been fully implemented yet. I did find that running Asterisk with the "-p" startup option helps somewhat. Have others found other cheap SIP/IAX hard-phones that behave better is this a systemic problem with the PA1688 phones? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On Sat, 30 Oct 2004 10:59:22 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On October 30, 2004 10:37 am, Benjamin on Asterisk Mailing Lists wrote: The X100P is not only an FXO device. Many folks use it as a Zaptel timing source only. Yes you are of course correct; but that then raises the question of why he wants two in there. Does he? Oh well, I guess you could have two for redundancy ;-) rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] confusing info from Digium and asteriskdoc about setup of TDM11B
I received my TDM11B as I assume everyone currently does with the green FXS module nearest the bracket (slot 1), and the red FXO card in the far slot (slot 4). After installing the card and the zaptel modules I believe I have the modules loaded correctly, modprobe wcfxs returns: Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not Installed Module 2: Not Installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) The instructions at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html tell me that I should use in my /etc/zaptel.conf: fxoks=1 # Make sure that the FXS(green) module is closest to the bracket... fxsks=4 # FXO module defaultzone=us loadzone=us But a direct note from Digium gave me the following (which I suspect has 2 instead of 4): - Hi sir, ALl the files will be the same as the online one, if you want the examples needed to make your cards work, append these to the following files /etc/ZAPTEL.CONF: fxoks=1 fxsks=2 /etc/asterisk/ZAPATA.CONF signalling=fxo_ks group=1 callerid=Steve (555)555- channel=1 signalling=fxs_ks group=2 callerid=asrecieved channel=2 -- As in the website docs, ztcfg returns: /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Now the website shows the following for /etc/asterisk/zapatel.conf, but shouldn't the second channel be 4 and not 2??? language=en context=default switchtype=national signalling=fxo_ks channel = 1 signalling=fxs_ks channel = 2 Now when I have an analog phone plugged into jack #1 on the back of the card and I start asterisk I get nothing but steady static over that phone. I would have expected nothing, or by some miracle a dialtone, but not steady static. Can anyone help get me started? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and codec
On Sat, 30 Oct 2004 17:35:51 +0200, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: the SIP doc on the wiki said that DTMF inband only worked on G.711. If I use G.726 and RFC2833 DTMF, will asterisk re-generate the DTMF at the Zap interface? I'm pretty sure it does. I am using SIP phones with DTMF set to SIP info and I call out on Zap devices all the time, no problem with DTMF. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info from Digium and asteriskdoc aboutsetup of TDM11B
- Original Message - From: Steve Prior [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, October 30, 2004 11:53 AM Subject: [Asterisk-Users] confusing info from Digium and asteriskdoc aboutsetup of TDM11B I received my TDM11B as I assume everyone currently does with the green FXS module nearest the bracket (slot 1), and the red FXO card in the far slot (slot 4). After installing the card and the zaptel modules I believe I have the modules loaded correctly, modprobe wcfxs returns: Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not Installed Module 2: Not Installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) The instructions at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html tell me that I should use in my /etc/zaptel.conf: fxoks=1 # Make sure that the FXS(green) module is closest to the bracket... fxsks=4 # FXO module defaultzone=us loadzone=us But a direct note from Digium gave me the following (which I suspect has 2 instead of 4): - Hi sir, ALl the files will be the same as the online one, if you want the examples needed to make your cards work, append these to the following files /etc/ZAPTEL.CONF: fxoks=1 fxsks=2 /etc/asterisk/ZAPATA.CONF signalling=fxo_ks group=1 callerid=Steve (555)555- channel=1 signalling=fxs_ks group=2 callerid=asrecieved channel=2 -- As in the website docs, ztcfg returns: /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Now the website shows the following for /etc/asterisk/zapatel.conf, but shouldn't the second channel be 4 and not 2??? language=en context=default switchtype=national signalling=fxo_ks channel = 1 signalling=fxs_ks channel = 2 Now when I have an analog phone plugged into jack #1 on the back of the card and I start asterisk I get nothing but steady static over that phone. I would have expected nothing, or by some miracle a dialtone, but not steady static. Can anyone help get me started? Steve Yes, it should be four unless you care to move the actual module on the card to the second slot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Modifying CDR data?
If you are in AGI... make your own call log. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Saturday, October 30, 2004 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Modifying CDR data? Roy, On Sat, 2004-10-30 at 15:31, Roy Sigurd Karlsbakk wrote: I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this? I don't think you can change dst from the extension flow just like that (maybe via an app, but that might have alternate consequences) I've done some scripting with entirely different purposes, but it may fit your needs: create an AGI script that is called when a call comes in, use that to store the uniqueid of the call leg into a database. Then check if call diversion is active and log that too. Afterwards, check (i.e. once an hour or whatever is convenient) and match CDR versus your own database. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info from Digium and asteriskdoc aboutsetup of TDM11B
On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro [EMAIL PROTECTED] wrote: Yes, it should be four unless you care to move the actual module on the card to the second slot. I have fixed this in CVS now. Should be propogated to the website in a few minutes. While we do try and test everything, sometimes things get missed. This is why getting people to test the configurations in Volume-One and report back what does and does not work is important. Thanks for pointing one out! Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
On Sat, 30 Oct 2004, Michael Graves wrote: On Sat, 30 Oct 2004 09:37:00 +0200 (CEST), Remco Barende wrote: Hi list! I found an interesting wireless phone product. Tiptel will be selling a base station for DECT phones that is VOIP capable. The base station comes in two models, one with the SCCP the other with H.323 protocol support. The interesting bit is that the base station is VOIP connected but you can use standard (cheap) DECT phones to connect to it. I found it on www.tiptel.nl (the websites in other countries do not mention this model) and the model is called tiptel DECT-Z 600 IP systeem Yu can find it on www.tiptel.nl - producten - DECT Draadloze telefoonsystemen - tiptel DECT-Z 600 IP systeem http://www.tiptel.nl/fr_top.asp?lang_id=1mid=901pid=901 Unfortunately the product page is in Dutch only. I wonder if Tiptel make this themselves or whether they bough an OEM product from a manufacturer. When googling I could not find a similar product though and there aren't nay mentioned under Wireless VOIP in the wiki either. Are there other similar base stations? Please forgive my ignorance, but what exactly is DECT? I don't see the term referenced anywhere in North America. Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 !michael.tag ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation
As one of the people who introduced both DECT and CT3 into the Australian enterprise market I'll take a crack at answering this. If anyone thinks this information is worthwhile I'll add it to the Wiki. DECT is a cordless phone solution. It can be sold as a stand alone single handset - single base station residential solution however it's main use is in a multiple base station multiple handset corporate solution. It is an evolution of the original proprietary CT3 technology developed by Ericsson in 1993. It was released as an open standard in 1996 and although some technical changes were made to CT3 to become DECT it is more or less the same. It is a TDMA style transmission with between 8-12 time slots per base station (vendor dependant). Generally each base station is powered locally and cabled back to a central point in a star configuration. You can generally get between 200 to 400 meters from the handset to the base station, though with the use of external yagi antennas you can get good directional connectivity for up to 1000m. Dect allows seamless handover and roaming between base stations, and although not widely published there are mobility location servers that allow you to use the same handset at multiple locations. Eg. I could be an area supervisor using my handset at one Home Depot and I travel to the next Home Depot and the mobility server will recognize that you are; A- at the new location B- tell the pabx to divert all of your calls to the new site. Whilst originally sold as a stand alone adjunct box to a pabx (generally using an E1 connection between the pabx and the DECT adjunct box) it has over the past 6 years evolved to where most PABX (European anyway) can connect the base stations directly to a chassis card installed directly in the PABX backplane. In 1998 the G.A.P. (Generic Access Profile) standard was published that allowed most handsets to operate with most base stations however it is generally a subset of the full proprietary feature list. (also known by the vendors as Go Away Please). There were a number of dual mode gsm/dect handsets developed over the years (I myself used to use a dual mode CT3/GSM Ericsson handset in 96 and later occasionally a Dual mode GSM/Dect Nokia handset) however they were not widely available and had a number of issues, generally battery life because most of them generally ran two internal radios simultaneously. There was of course no roaming between the dect and gsm carrier networks. One of the reasons you may not be familiar with DECT is that it was mainly a European standard (also NEC had a solution called PHS that was only ever available in Japan). Motorola and Lucent (or the Avaya division today) had a competing less capable solution called CT2 (less features, less time slots per base, less density per given area and also less roaming between bases stations). CT2 was also trialed as a poor mans cellular in Canada, 2 state based USA trials, and Hong Kong. Basically it allowed one way calling (eg handset to base only) so you could make outgoing calls but not receive calls. The Hong Kong solution was probably the most successful of these of these and interestingly they also had a built in pager for receiving text messages. There are no public access CT2 solutions currently operating. There are a number of interesting parallels to draw between the CT2 and DECT rollouts and the current Wifi Voip handsetsand let me tell you if you think there is nothing to learn from history then well you must know it all. Will WIFI Voip handsets replace desk based cabled handsets? Do end users really need combined VOIP/GSM handsets? Will carriers ever introduce mobility between enterprise VOIP and GSM? Where can enterprises benefit from a mobile solution? How can a carrier benefit from the changing technology to offer higher value add solutions in the enterprise market. Will voip wifi offer the ability for a small and nimble voip carrier to compete against the mobile gsm carriers. Which manufacturers are leading development and how do I find them. These and many other points are available to be learned by anyone who wants to talk to me about them. I'm now based in NY and can be reached via this email address. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Saturday, October 30, 2004 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation On Sat, 30 Oct 2004 09:37:00 +0200 (CEST), Remco Barende wrote: Hi list! I found an interesting wireless phone product. Tiptel will be selling a base station for DECT phones that is VOIP capable. The base station comes in two models, one with the SCCP the other with H.323 protocol support. The interesting bit is that the base station is VOIP connected but you can use standard (cheap) DECT phones to connect to it. I found it
Re: [Asterisk-Users] Is NuFone messing up for anybody else?
On Friday 29 October 2004 05:09 pm, Paul Rodan wrote: 1-way audio problems. At least I think it's one way. We hear the remote party breaking up. So it would be NuFone's ability to transmit, upload, or our download bandwidth. We're not having bandwidth issues, we have 4 DS3's at only about 70% of total capacity. I'm having the same issues. Though my call volume is really low. Using other servers than NuFone's I've not had the problem. There has been numerous problems making calls - with no line available. They say it's not them, but it does not help me if the route to them is to laggy or whatever the problem might be. Calling via a server (coast to coast) and I never have a problem, but often I do through NuFone. I know they often work late helping people but I've started a few threads that never went anywhere so I'm in a bit of a mystery as to what is going on. Does not make for a safe business plan. (I use a dedicated WAN pipe with QoS set on my Asterisk box.) Last week I upgraded from a CVS Head from around 9-15-04 to the CVS Stable on 10-26; However, no immediate problems were detected. Smooth upgrade. We place many calls a day and only today is it worst than usual. Just wanted to see if anybody else detected it, apparently not. Will look for another U.S termination provider with similar or better rates and move NuFone to secondary. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Chaney Sent: Friday, October 29, 2004 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is NuFone messing up for anybody else? Can you be more descriptive on what's happening? I use NuFone and haven't had any issues, but I don't make that many calls. On Fri, 29 Oct 2004 14:39:57 -0400, Paul Rodan [EMAIL PROTECTED] wrote: We've been using NuFone for about 2 months, pushing an average of 10,000 minutes a month of Long distance. There have been minor quality issues in the past, maybe to one area code or another. However, I've noticed today more problems than usual. Gotten several calls. Has anybody else noticed this? FYI. I'm running Asterisk CVS-v1-0-10/26/04-07:28:01 on Asterisk Server A, and Asterisk CVS-v1-0-10/25/04-16:04:29 on Asterisk Server B My phone connects to Asterisk Server B. Asterisk Server B connects to Asterisk Server A via IAX, which is off of the same switch. Asterisk Server A connects to NuFone via IAX. Cisco_7960 -SIP- Asterisk_B -IAX- Asterisk_A -IAX- NuFone via Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 190/220
On Friday 29 October 2004 12:32 pm, Ronald Hartmann wrote: Good Day list, I have spent better part of the morning reading through the user group messages and have found some people stating that they are able to get the Transfer Button to work on the Snom 190/220 Yes, press the softbutton that says Xfer (means transfer). -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On 06:57 AM 10/30/2004, Andrew Kohlsmith wrote: On October 29, 2004 11:49 pm, Chris A. Icide wrote: Only in the X100P format, and only 2 of them I have to ask -- why are you running such high-end equipment for a craptastic FXO device? Don't you find other issues that going to a TDM4xxP or even a T1+channel bank would fix? I mean I ran an X100P and TDM410P (1FXO+1FXS) on a weenie P90 for cryin' out loud... What's this super big system getting you (except maybe future proofing?) The box is one of my development boxes for developing Asterisk based solutions for my clients. I've always been of the mind that you can't have enough power... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is NuFone messing up for anybody else?
I have had the no line available a couple of times. But so far the call quality hasn't been bad, but I only make like 1 or 2 calls a day and they are short. On Sat, 30 Oct 2004 13:57:08 -0400, steve szmidt [EMAIL PROTECTED] wrote: On Friday 29 October 2004 05:09 pm, Paul Rodan wrote: 1-way audio problems. At least I think it's one way. We hear the remote party breaking up. So it would be NuFone's ability to transmit, upload, or our download bandwidth. We're not having bandwidth issues, we have 4 DS3's at only about 70% of total capacity. I'm having the same issues. Though my call volume is really low. Using other servers than NuFone's I've not had the problem. There has been numerous problems making calls - with no line available. They say it's not them, but it does not help me if the route to them is to laggy or whatever the problem might be. Calling via a server (coast to coast) and I never have a problem, but often I do through NuFone. I know they often work late helping people but I've started a few threads that never went anywhere so I'm in a bit of a mystery as to what is going on. Does not make for a safe business plan. (I use a dedicated WAN pipe with QoS set on my Asterisk box.) Last week I upgraded from a CVS Head from around 9-15-04 to the CVS Stable on 10-26; However, no immediate problems were detected. Smooth upgrade. We place many calls a day and only today is it worst than usual. Just wanted to see if anybody else detected it, apparently not. Will look for another U.S termination provider with similar or better rates and move NuFone to secondary. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Chaney Sent: Friday, October 29, 2004 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is NuFone messing up for anybody else? Can you be more descriptive on what's happening? I use NuFone and haven't had any issues, but I don't make that many calls. On Fri, 29 Oct 2004 14:39:57 -0400, Paul Rodan [EMAIL PROTECTED] wrote: We've been using NuFone for about 2 months, pushing an average of 10,000 minutes a month of Long distance. There have been minor quality issues in the past, maybe to one area code or another. However, I've noticed today more problems than usual. Gotten several calls. Has anybody else noticed this? FYI. I'm running Asterisk CVS-v1-0-10/26/04-07:28:01 on Asterisk Server A, and Asterisk CVS-v1-0-10/25/04-16:04:29 on Asterisk Server B My phone connects to Asterisk Server B. Asterisk Server B connects to Asterisk Server A via IAX, which is off of the same switch. Asterisk Server A connects to NuFone via IAX. Cisco_7960 -SIP- Asterisk_B -IAX- Asterisk_A -IAX- NuFone via Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 190/220
I understand about the soft button and yes this does work, however I am trying to figure out if the actual Button (the mechanical one on the phone) That says transfer is it possible to get this to work. ~ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lucent iMerge
I too am trying to get asterisk to connect to Lucent iMerge. Any chance of somebody sending me a ethereal trace of succesful connection to an iMerge? My conversation with it very short. I sent a gatekeeperRequest. The iMerge sends a gatekeeperReject. Two packets. That's it. I'd love to see what a gatekeeperRequest that works looks like. John -- John Gray gray at agora dash net dot com AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
On Sat, 30 Oct 2004, Remco Barende wrote: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. In short DECT is a kick-ass cordless phone system. Up to 8 phones, intercom calls between them etc etc etc. Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
[EMAIL PROTECTED] wrote: On Sat, 30 Oct 2004, Remco Barende wrote: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. In short DECT is a kick-ass cordless phone system. Up to 8 phones, intercom calls between them etc etc etc. Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so There are PCI DECT Boards but I don't know if they will do base-station stuff. *http://tinyurl.com/3hnq5 Maybe somebody else has more insight into this if those PCI and PCMCIA Dect boards can be turned into base-stations. -- Thomas * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation
No those cards are used to connect wirelessly to a base station. Basically like an 802.11x card connects to a base station. What Remco means is that he wants someone to release software that would sit on your pc. All a PCI card to connect to base stations and perform those functions. Remco, look into the Kirk h323 solution, this does what you need. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Gallaway Sent: Saturday, October 30, 2004 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation [EMAIL PROTECTED] wrote: On Sat, 30 Oct 2004, Remco Barende wrote: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. In short DECT is a kick-ass cordless phone system. Up to 8 phones, intercom calls between them etc etc etc. Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so There are PCI DECT Boards but I don't know if they will do base-station stuff. *http://tinyurl.com/3hnq5 Maybe somebody else has more insight into this if those PCI and PCMCIA Dect boards can be turned into base-stations. -- Thomas * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: AW: [Asterisk-Users] Firefly 1.9.6 released
We have had some problems registering the firefly with the Asterisk 1.0.2 it seams that IAX version doesn't match? How to solve this? Robert -Ursprungligt meddelande- Fran: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Pascal C. Kocher Skickat: den 28 oktober 2004 11:25 Till: Asterisk Users Mailing List - Non-Commercial Discussion Amne: AW: AW: [Asterisk-Users] Firefly 1.9.6 released Hello Tim I've found a bug in the new code that could have caused this problem. Try downloading and installing from http://www.virbiage.com/firefly/download/firefly-thirdparty.ex e again; this time you should get build 3941, which should solve the problem you ran into. Please let me know if it doesn't, or if you have any other trouble. This seemed to have fixed it, thank you very much! (4 hrs of operation so far) I'm really happy with Firefly, works like a charm ;) Thanks for the great work and for fixing it that fast! Best regards, Pascal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How far is IAX to be a Standard
Hello Guys Is Saturday and just to get my Actually poor Asterisk knowledge a little more rich How far is IAX to be a Industry Standard I mean like SIP or H.323 , IAX seems to be the answer to many many NAT problems in this Out-Of-Available-IP's World but what does the big guys (cisco , etc) think about IAX? what does the RFC's guys and the Pseudo-Cisco IETF think about this Protocol? I think This year is the consolidation of Asterisk is incredible. I abandon the track of asterisk for a while , but when I turn arround and see the advances I'm Back! Regards HA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How far is IAX to be a Standard
Hello Guys Is Saturday and just to get my Actually poor Asterisk knowledge a little more rich How far is IAX to be a Industry Standard I mean like SIP or H.323 , IAX seems to be the answer to many many NAT problems in this Out-Of-Available-IP's World but what does the big guys (cisco , etc) think about IAX? what does the RFC's guys and the Pseudo-Cisco IETF think about this Protocol? Who cares? It's not all that uncommon for a standard to be devised by people doing the work in an area, and then for the RFC to be written to document the standard. There are VoIP providers offering IAX, so it's clearly on the radar. If it's a worthy technology, it will eventually win out (and it looks to me like that's exactly what is happening). ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] moh
Hi, I have * 1.0.0. Everything works well except moh. I followed the instruction in http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the default mp3 from *. The problem is that the music is really slow. Seems like it didn't get the right rate to play. Any one having this problem too? Thanks, attachment: winmail.dat___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How far is IAX to be a Standard
On Sat, 30 Oct 2004 14:37:35 -0600, Voip Business [EMAIL PROTECTED] wrote: How far is IAX to be a Industry Standard I mean like SIP or H.323 Frank Miller, the author of the IAX specification document is doing further work on the specification with the intend to eventually submit to the IETF. He has been on other such IETF projects, so he know what he's doing, he's the right man for the job. However, we all know the realities of life, there are the things we would love to do, and there is the bread and butter business that takes up all of our time, leaving only very little time for the things we'd love to do. Frank, too is a busy man and therefore, please don't bother him about deadlines. If anybody wanted to speed this up I presume a sponsorship would be the kind of thing that could free up his time and thereby make it possible for him to spend more time on the draft. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 bandwidth efficiency calculations from Farfon
I thought this might be of interest to the list. http://www.convergence.com.pk/iax2/trunked.html Wasim, you should tell us about those things here. Anyway, keep up the good work ;-) rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
[EMAIL PROTECTED] schrieb: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so Someone does offer a DECT PCI-based base-station. Dosch Amand (http://www.dasystems.de/) have such a system. Unfortunately with Windows applications, only. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom IP 500/600
One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice delay with isdn
Hi all! I'm using asterisk (great!) with an hisax isdn card and it works well, but during normal phone calls (no voip) there is a delay between my voice going out and the the voice coming in... this delay increase (starting with 1 sec and going to 4-5 secs) with the time of the call. anyone help me? thx a lot. -- [fox] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo with long distance
Ok, I have an odd problem. I'm running Asterisk 1.0.1 and a setup like this: Analog phone -- Adtran channel bank -- Asterisk -- IAX -- Asterisk -- PRI For local calls it works perfectly. For long distance calls, it creates an echo. I don't have any problems when doing this with local or LD: SIP -- Asterisk -- PRI I have echo cancellation when bridged turn on on both Asterisk boxes. I need it for the one with the PRI, and have tried it both off and on with the one the channel bank is hooked up to. It's worse without the echo cancellation, but still there with it. My next attempt it to play with the number of taps a little. I don't want to drop the gain too far because I don't want people having to yell into the phone. Can you guys think of any reason it would pop up for LD but not for local calls? It might give me some further ideas on where to find the problem. Thanks, -- Mike Nugent [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] high-capacity systems / trouble with Tyan
On Fri, 29 Oct 2004, Michael Giagnocavo wrote: The only thing wrong with RedHat as far as asterisk is concerned is that they do something goofy with their kernels and all you need do is recompile a kernel from source. IMHO, you should always compile a kernel for your specific hardware. Does this mean that RHEL wouldn't really be a benefit to me? (Looking at getting a Dual Xeon Dell, and was thinking for $390 or so, it'd be nice to have it all supported, and supposedly RHEL's hyperthreading support is much better). I would suggest trying Tao Linux (http://www.taolinux.org). It is recompiled RHEL. It is what I use in production, totally stock kernels. No problems. Running T100Ps and TE405Ps. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 bandwidth efficiency calculations from Farfon
The chart is good, but I think it makes a mistake for iLBC: Isn't iLBC 13.something kbps? Also, since iLBC uses 30ms frames (when used with asterisk, at least), it has slightly lower overhead. Approx 2/3 as much overhead. (not that I'm a big iLBC fanboy or anything.. -- I still prefer a free codec). -SteveK On Oct 30, 2004, at 6:08 PM, Benjamin on Asterisk Mailing Lists wrote: I thought this might be of interest to the list. http://www.convergence.com.pk/iax2/trunked.html Wasim, you should tell us about those things here. Anyway, keep up the good work ;-) rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap, Highquality IP Phones
On Fri, 2004-10-15 at 22:54, James H. Thompson wrote: www.voxilla.com is usually one of the first places to get the new sipura products, at least this has been true in the past. Jim James H. Thompson [EMAIL PROTECTED] If you are in USA go ahead and try them but if you are outside of USA stay away from Voxilla! They don't know how to process Return/Repair shipments. We bought one SPA3000 sent it back to them as it was defective, when they return the unit back to us in Canada we ended up paying brokerage and taxes fee again as they don't know how to process Return shipment. Service is unprofessional. We try to contact them several times but they don't respond out our emails or calls. -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: asterisk SER and grandstream
In sip.conf, you should have the mailbox=MB# and host=ipofSER. And in ser.cfg, look for method==NOTIFY and do a t_relay() if in location. Don - Original Message - From: Yair Hakak [EMAIL PROTECTED] To: Asterisk Users List [EMAIL PROTECTED] Sent: Saturday, October 30, 2004 10:44 AM Subject: [Asterisk-Users] re: asterisk SER and grandstream hi list, anyone have any success getting asterisk to pass message waiting indicator to a grandstream with SER in the middle as a SIP proxy? I recently implemented SER between asterisk and my SIP clients and it's significantly more stable (no more dropped clients) but i haven't been able to figure out how to send message waiting indication so the grandstream's LCD flashes. If anyone has succesfully done this i'd be grateful for the info. thanks, yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom IP 500/600
The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c:7325
Hi, I've searched everywhere but I have not been able to get an answer to the following problem: I get the following notice (appropriate parts are taken out for security purposes) after long periods of registration using an spa 2000 with registration set to 3600 seconds and a proxy failover set to 20 seconds. I use version 2.08 as it has stayed registered longer than the current version 2.10(e). Also, I have tried a few spa2000's so I know it's not related to any particular box that has gone bad. Oct 31 00:01:01 NOTICE[-1116333136]: chan_sip.c:7325 handle_request: Registration from 'snip sip:snip@ip' failed for 'user ip' The box stays registered for a random period of time before resulting in the above message. It remains unregistered for another random period of time before becoming registered again. And the cycle continues. Any help would be very much appreciated. Regards, Mohammed Salim EZZI Telecom, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation
On Sat, 30 Oct 2004, dean collins wrote: No those cards are used to connect wirelessly to a base station. Basically like an 802.11x card connects to a base station. What Remco means is that he wants someone to release software that would sit on your pc. All a PCI card to connect to base stations and perform those functions. Remco, look into the Kirk h323 solution, this does what you need. Thanks for the excellent explanation on DECT! I will probably order the base station, it seems like an almost ideal solution to connect phones to a voip pabx. I would not prefer a pci card solution personally, anything connected to the network doesn't cause irq headaches :) I'll probably buy the Tiptel version of the Kirk because it's available locally (unless the price difference is huge), but they have two models supporting different protocols. Which protocol would be better h323 or SCCP to connect to *? There are some features described on the website specific to SCCP model: * Caller hold (only with Cisco Call Manager Express) * Support for Cisco Call Manager versie 3.3 (3) * Support for Cisco SRST (Survivable Remote Site Telefony) 3.0 * Support for Cisco Call Manager Express versie 3.0 Not sure whether I would really need it and I don't understand the part about caller on hold feature. Surely they can't be serious that you cannot put a caller on hold and transfer the call on the h323 version? I'll put that question to them on monday. Thanks!! Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Gallaway Sent: Saturday, October 30, 2004 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation [EMAIL PROTECTED] wrote: On Sat, 30 Oct 2004, Remco Barende wrote: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. In short DECT is a kick-ass cordless phone system. Up to 8 phones, intercom calls between them etc etc etc. Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so There are PCI DECT Boards but I don't know if they will do base-station stuff. *http://tinyurl.com/3hnq5 Maybe somebody else has more insight into this if those PCI and PCMCIA Dect boards can be turned into base-stations. -- Thomas * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom IP 500/600
My bad... I thought he was attempting to upload config files for asterisk systems. Yes, an expect script would work just fine... At 11:21 PM 10/30/2004, you wrote: The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersBest Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do I *need* to compile zaptel?
This is how I do it - I know the 2.6 kernel is supposed to have an easier way, but I've not seen/read how to do it yet. That did it for CVS head on a knoppix distro. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info from Digium and asteriskdoc aboutsetup of TDM11B
Looks like it's still incorrect in the first blue paragraph of the section on FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that section twice still calls the channel # 2. Now on to my next confusion... The section on contexts under dislplans mentions a context named [incoming]. This isn't a context that's mentioned anywhere before this and it's not at all clear where it comes from - I'm starting to suspect that some context references belong in the zapatel.conf file. A comment about where the document leaves off. In the beginning the document promises to get to a minimal working set, but it really doesn't go that far. Unless I've missed something, we aren't left with even a complete version of the minimal example extensions.conf file. Something is missing so that I'm not getting a dial tone on the analog phone hooked up to the TDM11B and I have no idea why (can anyone clue me in?) I also tried the: [incoming] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() example and asterisk didn't appear to see the incoming call and answer the call at all. I'd love for the example files to be complete enough that this example could actually work from either the external POTS line or even better an analog phone hooked to the FXS interface. I think it would be great if attached to the document there was a final version of all of the config files which are known to work with the given configuration. Can you help get me to a dialtone on the internal side or an answer on the external side? Thanks Steve Leif Madsen wrote: On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro [EMAIL PROTECTED] wrote: Yes, it should be four unless you care to move the actual module on the card to the second slot. I have fixed this in CVS now. Should be propogated to the website in a few minutes. While we do try and test everything, sometimes things get missed. This is why getting people to test the configurations in Volume-One and report back what does and does not work is important. Thanks for pointing one out! Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] polycom IP 500/600
If the phone is behind a NAT firewall, it would require extra configuration on the firewall. Depending on the circumstance, it is not always be possible to make such a change. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Saturday, October 30, 2004 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] polycom IP 500/600 My bad... I thought he was attempting to upload config files for asterisk systems. Yes, an expect script would work just fine... At 11:21 PM 10/30/2004, you wrote: The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
