[asterisk-users] Strange error, logger.c: No more room in scheduler...
I found no info about this strange error: logger.c: No more room in scheduler logger.c: Asked to delete sched id -1??? Only in verbose mode. Someone know how to solve this? Asterisk 1.2.13 with sangoma A104EC Hints? Thnks. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
On Mon, Apr 09, 2007 at 10:43:27PM -0700, [EMAIL PROTECTED] wrote: I have a Digium TDM400b11, 1FXO [port2] 1FXS [port 1] When I reload the chan_zap I get: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be specified before any channels are. Apr 9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Apr 9 22:39:36 WARNING[3541]: loader.c:554 load_modules: Loading module chan_zap.so failed! Here is my zapata.conf cat /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ; ;=== ; define the channels signaling=fxo_ks signalling=fxo_ks (note the double 'l') context=internal channel = 1 signaling=fxs_ks context=incoming channel = 2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: G729 'disappears' randomly
That's what it was... I should have posted :-) playing with /etc/mactab and nameif to fix it. -r On 4/7/07, Nikolai Lusan [EMAIL PROTECTED] wrote: On Fri, 2007-03-23 at 03:11 +0530, Rajeev Natarajan wrote: It happened again this evening and when I checked the host-id in /var/log/asterisk/messages the time when it did not register, it showed a host-id Mar 22 18:14:48 VERBOSE[2586] logger.c: == G.729 Host-ID: 90:23:3a:b7:dc:46:88:fc:cf:bb:78:a2:b8:00:75:97:34:xx:xx:xx (removing the last 6 for security) and it did not load the g729 Mar 22 18:43:18 VERBOSE[2580] logger.c: == G.729 Host-ID: 05:e5:4b:6c:0d:8b:66:fd:7a:b5:8e:a6:23:73:0b:b1:66:xx:xx:xx WORKS perfectly Any clues on why the host-id changes? IDEA: I also notice that sometimes eth0,eth1 and eth2 (Yes: i have three network interfaces) interchange on reboot. Are they related? Quite possibly, the registration program for that codec will bind to eth0 and use it as the host ID, if you change ethernetcards or re-number interfaces you will need to re-register the codec. As for the re-ordering of your network cards I would suggest you look into running udev with some rules to keep the order of the cards consistent over reboots. -- Nikolai Lusan # # # Weblog: http://lusan.id.au/~nikolai/blog # Website:http://lusan.id.au/~nikolai # # ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
These are the patent numbers in the lawsuit (Thanks Pat and Sal) 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 Mark C. Yuan LIU wrote: From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QSIG configuration
Hello, Anyone to help with information on configuring Q.SIG in Asterisk ? I run ASTERISK 1.4 with a Wildcard TE410P-Xilinx on Fedora Core 6 and Zaptel 1.4 I need to tie this ASTERISK system to a Panasonic TDA200 PABX which has ISDN PRI Card which requires QSIG signalling for seamless integration with the ASTERISK system. Any help would be appreciated. Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip_header=value?
I have tried your suggestion but it doesnt work, here is my extensions.conf:
exten= 123,1,Set(__SIPADDHEADER=Call-Info:answer-after=0);
exten= 123,2,Dial(SIP/abc/${EXTEN},,Tt)
exten= 123,3,hangup
the cli displays busy or congested message if sipura is registered with my
asterisk, and if sipura is registered to another asterisk and that asterisk
is registered as a peer to my asterisk then it rings the phone. Plz help
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Apparently it sets a SIP_HEADER variable named Call-Info to a value of
answer-after=0 effectively telling the Sipura to answer the call and put
it through to speakerphone.
I will say that extensions.ael is a bit different from regular line based
extensions.conf in that I seem to have to escape all sorts of stuff with
the \ character that I don't have to in extensions.conf
Back to work, I'll check in on this thread later this evening.
Rizwan Hisham wrote:
I dont understand it
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
whats it doing here?
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael
Rizwan Hisham wrote:
I have tried it, it doesnt work
On 4/9/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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[asterisk-users] Autoreply: [Posible Spam] asterisk-users Digest, Vol 33, Issue 36
Su mensaje fué reenviado a un técnico de Mildmac. Para consultas técnicas, por favor, envíe sus mensajes a [EMAIL PROTECTED] Gracias. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] checking credit by phone
Hi to all, I've tried to use the ASTCC credit check a long time ago and it worked pefectly, but now... no more Any suggestions for some new software? Thanks to all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zaptel 1.4.1 Install Modules CentOS
Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36
So [EMAIL PROTECTED] gets back to his office tomorrow. Any thoughts on how we should welcome him back? :) Think nice not nasty. I was thinking everyone from the list should send him a welcome back email in the next 24 hours. Or and I have no idea where this started but I got it from another list I subscribe to, if you screw up there (and I don't mean auto-responder either I mean if you saying something dumb or the like) they all send you photos of snow. Nothing interesting just endless photos of Snow. Any other suggestions? Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 10 April 2007 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36 Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan help - MeetMe and call monitoring
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a callcenter)
The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its
telephone should ring when he needs to listen to a call.
I was thining at using a MeetMe in which i'd put both legs of the
monitored call and the person who should hear the conversation.
Do you have other tips about that ??
Here was my first idea of dialplan to get to it.
[outgoing]
exten = _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten = _X.,n,Answer()
exten = _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = _X.,n,Wait(1)
exten = _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s^1))
[invite-third-party]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s^1))
[bridge-all]
exten = s,1,MeetMe(${MEETMEROOM},qdx)
exten = s,2,MeetMe(${MEETMEROOM},mqdx)
This setup is not working because I cannot call a Dial again on a
bridged channel
Here is what I get on Asterisk CLI
== Starting SIP/180-108c94b0 at call-third-party,s,2 failed so
falling back to exten 's'
== Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so
falling back to context 'default'
Do you have some idea to achieve this kind of result ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
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Re: [asterisk-users] Dialplan help - MeetMe and call monitoring
Edoardo Serra wrote:
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a
callcenter)
The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its
telephone should ring when he needs to listen to a call.
I was thining at using a MeetMe in which i'd put both legs of the
monitored call and the person who should hear the conversation.
Do you have other tips about that ??
You could use a Local Channel, one leg recording to a soundfile, the
other for the conference room.
Here was my first idea of dialplan to get to it.
[outgoing]
exten = _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten = _X.,n,Answer()
exten = _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = _X.,n,Wait(1)
exten = _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s^1))
[invite-third-party]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s^1))
[bridge-all]
exten = s,1,MeetMe(${MEETMEROOM},qdx)
exten = s,2,MeetMe(${MEETMEROOM},mqdx)
This setup is not working because I cannot call a Dial again on a
bridged channel
Here is what I get on Asterisk CLI
== Starting SIP/180-108c94b0 at call-third-party,s,2 failed so
falling back to exten 's'
== Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so
falling back to context 'default'
Do you have some idea to achieve this kind of result ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
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Knud A. Müller
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RE: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36
On Tue, 10 Apr 2007, Dean Collins wrote: So [EMAIL PROTECTED] gets back to his office tomorrow. Any thoughts on how we should welcome him back? :) Think nice not nasty. I was thinking everyone from the list should send him a welcome back email in the next 24 hours. Or and I have no idea where this started but I got it from another list I subscribe to, if you screw up there (and I don't mean auto-responder either I mean if you saying something dumb or the like) they all send you photos of snow. Nothing interesting just endless photos of Snow. Any other suggestions? A short WAV of the number unobtainable tone from your lcoal telco ... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip_header=value?
I have discovered another thing,
exten= 123,1,Dial(SIP/abc/${EXTEN},,Tt)
exten= 123,2,hangup
If the user is using xlite to register with my asterisk server, then we are
able to call him using the above axtension, and if the user is using sipura
the the above extension does not dial the user instead it displays a
congestion message as before, maybe there is a problem in sipura firmware. I
am using Linksys/SPA2100-3.3.6. any ideas why is sipra behaving like this.
for sipura to ring we have to use the following extension, without ${EXTEN}
variable
Dial(SIP/abc,,Tt)
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Apparently it sets a SIP_HEADER variable named Call-Info to a value of
answer-after=0 effectively telling the Sipura to answer the call and put
it through to speakerphone.
I will say that extensions.ael is a bit different from regular line based
extensions.conf in that I seem to have to escape all sorts of stuff with
the \ character that I don't have to in extensions.conf
Back to work, I'll check in on this thread later this evening.
Rizwan Hisham wrote:
I dont understand it
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
whats it doing here?
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael
Rizwan Hisham wrote:
I have tried it, it doesnt work
On 4/9/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Rizwan Hisham
Software Engineer
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[asterisk-users] Dialplan help - MeetMe (or ChannelRedirect) and call monitoring
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a
callcenter)
The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its
telephone should ring when he needs to listen to a call.
I was thining at using a MeetMe in which i'd put both legs of the
monitored call and the person who should hear the conversation.
Do you have other tips about that ??
Here was my first idea of dialplan to get to it.
[outgoing]
exten = _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten = _X.,n,Answer()
exten = _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = _X.,n,Wait(1)
exten = _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s1))
[invite-third-party]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s1))
[bridge-all]
exten = s,1,MeetMe(${MEETMEROOM},qdx)
exten = s,2,MeetMe(${MEETMEROOM},mqdx)
This setup is not working because I cannot call a Dial again on a
bridged channel
Here is what I get on Asterisk CLI
== Starting SIP/180-108c94b0 at call-third-party,s,2 failed so falling
back to exten 's'
== Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so
falling back to context 'default'
Do you have some idea to achieve this kind of result ?
Maybe I can use ChannelRedirect from Asterisk 1.4 ?
Cna you give me a hint on that ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino
Tel: +39 011 678 100
Fax: +39 011 678 275
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Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36
Everybody could send him a message saying I am not out of the office Dean Collins wrote: So [EMAIL PROTECTED] gets back to his office tomorrow. Any thoughts on how we should welcome him back? :) Think nice not nasty. I was thinking everyone from the list should send him a welcome back email in the next 24 hours. Or and I have no idea where this started but I got it from another list I subscribe to, if you screw up there (and I don't mean auto-responder either I mean if you saying something dumb or the like) they all send you photos of snow. Nothing interesting just endless photos of Snow. Any other suggestions? Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 10 April 2007 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 36 Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip_header=value?
There is a difference bwtween the SPA-941 which is a standalone desktop
phone w/ Speakerphone, and your SPA2100
What header value are you trying to set, and why?
Because the example I gave is for making the SPA-941 an intercom, and
also requires some changes in it's settings within the phones web
interface. So of course my example isn't going to anything for your
SPA-2100 except give you an example of how to set a SIP header.
So now my question becomes, What is it that you are attempting to do by
setting the SIP Header?
Best Regards,
Karl J. Vesterling
Rizwan Hisham wrote:
I have discovered another thing,
exten= 123,1,Dial(SIP/abc/${EXTEN},,Tt)
exten= 123,2,hangup
If the user is using xlite to register with my asterisk server, then
we are able to call him using the above axtension, and if the user is
using sipura the the above extension does not dial the user instead it
displays a congestion message as before, maybe there is a problem in
sipura firmware. I am using Linksys/SPA2100- 3.3.6. any ideas why is
sipra behaving like this.
for sipura to ring we have to use the following extension, without
${EXTEN} variable
Dial(SIP/abc,,Tt)
On 4/9/07, *Karl J. Vesterling* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Apparently it sets a SIP_HEADER variable named Call-Info to a
value of answer-after=0 effectively telling the Sipura to answer
the call and put it through to speakerphone.
I will say that extensions.ael is a bit different from regular
line based extensions.conf in that I seem to have to escape all
sorts of stuff with the \ character that I don't have to in
extensions.conf
Back to work, I'll check in on this thread later this evening.
Rizwan Hisham wrote:
I dont understand it
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
whats it doing here?
On 4/9/07, *Karl J. Vesterling* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael
Rizwan Hisham wrote:
I have tried it, it doesnt work
On 4/9/07, *Hermann Wecke* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i
like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Rizwan Hisham
Software Engineer
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[asterisk-users] clarification about bridge the call
Hello all, we will bridge the two asterisk call, after that i am trying to redirect the call to ivr. here i faced some problem 1)originate the first call sip/1-234 2)originate second call sip/2-245 3)bridge both the call 4)redirct both the call to IVR the call has been hangup. which cammand i have to use in asterisk manager API Regards, Pandi.P ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
On Tue, 10 Apr 2007 09:46:25 +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: Here is my zapata.conf cat /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ; ;=== ; define the channels signaling=fxo_ks signalling=fxo_ks (note the double 'l') context=internal channel = 1 signaling=fxs_ks context=incoming channel = 2 I still get: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 10 06:48:00 ERROR[4863]: chan_zap.c:10388 setup_zap: Signalling must be specified before any channels are. Apr 10 06:48:00 WARNING[4863]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Apr 10 06:48:00 WARNING[4863]: loader.c:554 load_modules: Loading module chan_zap.so failed! cat /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ; ;=== ; define the channels signaling=fxo_ks context=internal channel = 1 signaling=fxs_ks context=incoming channel = 2 -- Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with Sipura SPA 3000
Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reverse-ATA : Using PSTN lines to connect to Asterisk
Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone My own Asterisk - Outside lines But when it comes to smaller villages (I deal with people in tiny places), I'd like to reuse their own PSTN line this way: Customer's SIP hardphone My own Asterisk -- Some device on the customer's premise customer's PSTN lines I know ATAs are mostly used in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk without PSTN interface cards
People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards, Alejandro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect to Asterisk
On 4/10/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone My own Asterisk - Outside lines But when it comes to smaller villages (I deal with people in tiny places), I'd like to reuse their own PSTN line this way: Customer's SIP hardphone My own Asterisk -- Some device on the customer's premise customer's PSTN lines I know ATAs are mostly used in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike Hello Mike, Wouldn't a Sipura SPA 3000, with an FXS and an FXO, handle what you want? Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
And, sorry if I misunderstood, but you still have signaling, not signalling, in your zapata.conf? -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] maximum simultaneous calls
On 3/30/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Mark Quitoriano [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 23:05:57 +0800 Hi, what could be the maximum simultaneous calls can asterisk do? i read about the asterisk business edition review[1] and it can only handle 120 simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or less 90 simultaneous calls. [1] http://www.voiptalk.org/products/Asterisk+Business+Edition What about http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning ? People reports all kinds of numbers above 120. The answer partially depends on your hardware. Simultaneous calls can also mean very different things under different circumstances, as the page will tell you. If there is no transcoding/NAT'ing/in-band signaling, simultaneous calls can mean SIP set-ups only. You can see extremely high numbers even on ancient equipment. If everything is in-band and you are using CPU-intensive CODECs, the number will drop sharply. It also varies with types of channels, i.e., whether you use PSTN, IAX, SIP, H.323. But still, I don't think 120 is any limit. Yuan Liu Hi thanks for the reply. im just doing sip to sip all g729a codecs, my hardware is a dual xeon 3.0GHZ with 2gb mem. how many simultaneous calls can you estimate with this setup? Thanks! Mark Quitoriano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
On 4/10/07, Alejandro Cabrera Obed [EMAIL PROTECTED] wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards, Hola Alejandro, I used asterisk some days on a mac without any PSTN whatsoever, just to talk between softphones (and ip phones). No problem with that. Cheers, Francis Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange error, logger.c: No more room in scheduler...
Does this occur in the latest 1.2.17 release? On 4/10/07, Massimo Nuvoli [EMAIL PROTECTED] wrote: I found no info about this strange error: logger.c: No more room in scheduler logger.c: Asked to delete sched id -1??? Only in verbose mode. Someone know how to solve this? Asterisk 1.2.13 with sangoma A104EC Hints? Thnks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
Alejandro, On Tue, 10 Apr 2007, Alejandro Cabrera Obed said something to this effect: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? I am not sure if the debian release yet contains 1.4.x. It may be safer to compile from source if you want the bleeding-edge version. But 1.2.x works quite well, too, so it's really up to you. I prefer 1.4.x personally at this point, now that it's been hanging out there for a while. 2) Do I have to configure a dummy PSTN interface in my case ?? I am not sure exactly what you mean by dummy PSTN interface, but in principle, the answer is no. However, there are situations in which it may benefit you to have a Digium-derived card in the machine so that the RTC (real-time clock) on it can be used for driving the intervals of hold music, conference calling (MeetMe), etc. But there is also a ztdummy module that can be loaded into the kernel which emulates such an RTC. So, if your intent is purely to use SIP internally, no need for any PSTN. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect to Asterisk
Hi Mike, You should be looking at ATAs that have FXO, rather than FXS interfaces. Most ATAs come with FXS ports so that you can connect analogue phones to them, but in this case you're wanting to take PSTN lines from the outside, so FXO is desirable. Second, you'd have to make sure that the ATA supports the sort of application you're using it for; most are manufactured on the opposite premise. I am actually not sure offhand of any ATA firmware that I know that I imagine would work this way, although I'm confident it exists as consecutive back-to-back analogue-VoIP adaptations in many scenarios can get quite complex and requires that flexibility. Basically, you're looking for a small IP PBX that uses SIP internally among its private nodes and takes PSTN trunks from the outside. That's what PBXs typically do. :-) If all else fails, you can always roll your own functionality of this nature by using FXO cards in Asterisk. There are various distributions that package it in a very lightweight and reusable manner specifically for this type of purpose, or you can roll your own if it's scalable enough. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
On Tue, 10 Apr 2007, Alejandro Cabrera Obed wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. I run Debian servers. I also run Asterisk on them. In my case, I do not use the supplied Debian packags, but prefer to compile my own. That way I get the version of asterisk that I want with the config files where I want them (which oddly enough are as per the asterisk documentation) You only need a timing interface for some operations - MeetMe and possibly Music On Hold springs to mind, in which case, the ztdummy module will probably be sufficient for your needs. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 37
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
Probably, if I only needed one FXO. What is the customer has 4 channels (PSTN lines)? Don't I need 4 FXO? And, about the Sipura, it looks like it would do what I want, but it only has one FXO, limiting it's usefulness. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francis Augusto Medeiros Sent: Tuesday, April 10, 2007 10:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk On 4/10/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone My own Asterisk - Outside lines But when it comes to smaller villages (I deal with people in tiny places), I'd like to reuse their own PSTN line this way: Customer's SIP hardphone My own Asterisk -- Some device on the customer's premise customer's PSTN lines I know ATAs are mostly used in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike Hello Mike, Wouldn't a Sipura SPA 3000, with an FXS and an FXO, handle what you want? Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
On Tue, 10 Apr 2007 10:17:27 -0400 (EDT) Alex Balashov [EMAIL PROTECTED] wrote: And, sorry if I misunderstood, but you still have signaling, not signalling, in your zapata.conf? [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 10 07:35:44 WARNING[4960]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Apr 10 07:35:44 ERROR[4960]: chan_zap.c:7034 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 10 07:35:44 ERROR[4960]: chan_zap.c:10468 setup_zap: Unable to register channel '1' Apr 10 07:35:44 WARNING[4960]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Apr 10 07:35:44 WARNING[4960]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ; ;=== ; define the channels signalling=fxo_ks context=internal channel = 1 signalling=fxs_ks context=incoming channel = 2 [EMAIL PROTECTED] ~]# -- Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail: How to send a notification if Caller hags up during announcement
Hi, I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] maximum simultaneous calls
On 3/29/07, Matthew J. Roth [EMAIL PROTECTED] wrote: We are regularly running 250-300 simultaneous calls in an inbound call center environment. We had stability issues for a long time, but using weights on the queues was the needle in the haystack that was causing our problems. After removing them, our only failure in the past month has been a single segmentation fault. Were your problems limited to queue weights, or are agent penalties likely to cause this as well? I'm about to embark down the ACD path, and still unsure as to whether AgentCallbackLogin (on 1.2) or AddQueueMember (on 1.4) is the better solution to go with. Thanks -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Coccimiglio Sent: Tuesday, April 10, 2007 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit These are the patent numbers in the lawsuit (Thanks Pat and Sal) 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 Mark C. Yuan LIU wrote: From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Christopher Chan wrote: Just curious, Christopher, what is a chicken boner? Sorry, that's anti-spammer jargon for spammer. I used to be a mail admin for an outfit that handles over 40 million mailboxes and over 200 million email transactions daily. Guess what composed the majority of the daily 200 million transactions...and their origin...and the 'software' used http://www.spamfaq.net/terminology.shtml#chickenboner http://www.netlingo.com/lookup.cfm?term=chicken-boner ___ Thanks for the answer. I've never heard that one before. I remember once I used a 3rd party component set (Indy 9) to do some smtp alert emails from a Windows application. Couldn't get the mails to go through for a few particular customers and after some research and talking to ISP's, we found out that the component set that I used was used to write spamming software and the headers produced by the component set were flagged by spamm assasin and others. Arg. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble recording calls
anyone, anyone?
ahester wrote:
Hi all,
I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see got hangup request on the cli and
then mixmonitor ends.) I am guessing this has to do with the announce
file that is played to the engineer before the call is connected. It
seems that if the call rolls to the next queue because of timeout,
asterisk doesn't even try to record it. (I don't see any mixmonitor on
the cli for the next queue).
I would appreciate any help with this. I have to have all calls
recorded and I have to do announcements so that the callee knows how to
answer the phone.
Thanks,
Andy
The configs are as below:
From extensions.conf:
#after various menu stuff, send to support
exten = 214xxx,13,SetGlobalVar(ORIGIN=support)
exten = 214xxx,14,Queue(support1|tr|||10)
exten = 214xxx,15,Queue(support2|tr|||)
#dial command for sip extensions that are in the queues
exten =
_72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _72XXX,2,Dial(SIP/${EXTEN})
exten =
_73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _73XXX,2,Dial(SIP/${EXTEN})
queues from queues.conf:
[support1]
; Support call queue
announce = 16
strategy = rrmemory
timeout = 15
retry = none
wrapuptime=15
announce-frequency = 0
joinempty = no
leavewhenempty = yes
member = Agent/2008
member = Agent/2009
member = Agent/2014
member = Agent/2015
member = Agent/2017
member = Agent/2018
member = Agent/2019
member = Agent/3520
member = Agent/3521
member = Agent/3522
member = Agent/3524
member = Agent/3529
[support2]
; Support2 call queue
announce = 16
strategy = ringall
announce-frequency = 0
; Added below for testing because the second queue was not even trying
to record
; according to the asterisk console (still doesn't)
Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
monitor-format = wav
monitor-join = yes
joinempty = yes
member = SIP/72008
member = SIP/72009
--
Andy Hester
Network Engineer
Architel
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RE: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
Thanks Alex, That was my original thought, to just buy a TDM400 from Digium and put in as many FXO as I wanted, but I liked having the ease of just buying something off the shelf, even if it meant paying a little more. But it looks like I won't have much of a choice. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Tuesday, April 10, 2007 10:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk Hi Mike, You should be looking at ATAs that have FXO, rather than FXS interfaces. Most ATAs come with FXS ports so that you can connect analogue phones to them, but in this case you're wanting to take PSTN lines from the outside, so FXO is desirable. Second, you'd have to make sure that the ATA supports the sort of application you're using it for; most are manufactured on the opposite premise. I am actually not sure offhand of any ATA firmware that I know that I imagine would work this way, although I'm confident it exists as consecutive back-to-back analogue-VoIP adaptations in many scenarios can get quite complex and requires that flexibility. Basically, you're looking for a small IP PBX that uses SIP internally among its private nodes and takes PSTN trunks from the outside. That's what PBXs typically do. :-) If all else fails, you can always roll your own functionality of this nature by using FXO cards in Asterisk. There are various distributions that package it in a very lightweight and reusable manner specifically for this type of purpose, or you can roll your own if it's scalable enough. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
On Tue, Apr 10, 2007 at 10:28:46AM -0400, Mike said: Probably, if I only needed one FXO. What is the customer has 4 channels (PSTN lines)? Don't I need 4 FXO? And, about the Sipura, it looks like it would do what I want, but it only has one FXO, limiting it's usefulness. I strongly recommend that you check out the wiki: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon Vonage 101
These are apparently the patents involved. I cut and paste this info from another site. Vonage Infringed: Patent #6,282,574: http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1Sect2=HITOFFd=PALLp=1 u=%2Fnetahtml%2FPTO%2Fsrchnum.htmr=1f=Gl=50s1=6282574.PN.OS=PN/6282574 RS=PN/6282574 Patent #6,104,711: http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1Sect2=HITOFFd=PALLp=1 u=%2Fnetahtml%2FPTO%2Fsrchnum.htmr=1f=Gl=50s1=6104711.PN.OS=PN/6104711 RS=PN/6104711 Vonage Infringed, although not willfully: Patent #6,359,880: http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1Sect2=HITOFFd=PALLp=1 u=%2Fnetahtml%2FPTO%2Fsrchnum.htmr=1f=Gl=50s1=6359880.PN.OS=PN/6359880 RS=PN/6359880 No Infringement: Patent #6,137,869: http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1Sect2=HITOFFd=PALLp=1 u=%2Fnetahtml%2FPTO%2Fsrchnum.htmr=1f=Gl=50s1=6137869.PN.OS=PN/6137869 RS=PN/6137869 Patent #6,430,275: http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1Sect2=HITOFFd=PALLp=1 u=%2Fnetahtml%2FPTO%2Fsrchnum.htmr=1f=Gl=50s1=6430275.PN.OS=PN/6430275 RS=PN/6430275 -Original Message- From: Salvatore Giudice [mailto:[EMAIL PROTECTED] Sent: Saturday, April 07, 2007 11:46 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Verizon Vonage 101 Here's links and descriptions for the 8 you listed. All Bell Atlantic, GTE, or Verizon. This should make your research a bit easier. 6,137,869 Network session management http://www.google.com/patents?vid=USPAT6137869id=yl4GEBAJdq=6137869 Patent number: 6137869 Filing date: Sep 16, 1997 Issue date: Oct 24, 2000 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Network Services, Inc. Primary Examiner: Rexford N Barnie 6,430,275 Enhanced signaling for terminating resource http://www.google.com/patents?vid=USPAT6430275id=NmwLEBAJdq=6,430,275 Patent number: 6430275 Filing date: Jul 28, 1999 Issue date: Aug 6, 2002 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Services Network, Inc. Primary Examiner: Curtis Kuntz Secondary Examiner: Rexford M Barnie 6,104,711 (The famous: We think we invented ENUM patent) Enhanced internet domain name server http://www.google.com/patents?vid=USPAT6104711id=J18EEBAJdq=6,104,711 Patent number: 6104711 Filing date: Mar 6, 1997 Issue date: Aug 15, 2000 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,282,574 Method, server and telecommunications system for name translation on a conditional basis and/orto a telephone number http://www.google.com/patents?vid=USPAT6282574id=46sIEBAJdq=6,282,574 Patent number: 6282574 Filing date: Feb 24, 2000 Issue date: Aug 28, 2001 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,359,880 Public wireless/cordless internet gateway http://www.google.com/patents?vid=USPAT6359880id=tP4KEBAJdq=6,359,880 Patent number: 6359880 Filing date: Jul 30, 1999 Issue date: Mar 19, 2002 Inventors: James E. Curry, Robert D. Farris Primary Examiner: Wellington Chin Secondary Examiner: Steven Nguyen 6,128,304 (We think we own presence too...) Network presence for a communications system operating over a computer network http://www.google.com/patents?vid=USPAT6128304id=BnkGEBAJdq=6,128,304 Patent number: 6128304 Filing date: Oct 23, 1998 Issue date: Oct 3, 2000 Inventors: Steven E. Gardell, Barbara Mayne Kelly, Rajiv Bhatnagar, Thomas James Antell, Israel B. Zibman Assignee: GTE Laboratories Incorporated Primary Examiner: Frank Duong 6,298,062 (aka. Accepting H.323 phone calls/faxes from a computer network and terminating them on the PSTN) System providing integrated services over a computer network http://www.google.com/patents?vid=USPAT6298062id=jp4IEBAJdq=6,298,062 Patent number: 6298062 Filing date: Oct 23, 1998 Issue date: Oct 2, 2001 Inventors: Steven E. Gardell, Israel B. Zibman Assignee: Verizon Laboratories Inc. Primary Examiner: Shick Hom -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Buller Sent: Saturday, April 07, 2007 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon Vonage 101 J. Oquendo wrote: So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... I am looking at Verizon Services Corp. v. Vonage Holdings Corp., Slip Copy, 2007 WL 528749, E.D.Va.,2007, which is the result of the Markman hearing. That is the court interpreting the claim language, and here are the patents discussed: 6,137,869
Re: [asterisk-users] maximum simultaneous calls
On Tue, 10 Apr 2007, James FitzGibbon wrote: I'm about to embark down the ACD path, and still unsure as to whether AgentCallbackLogin (on 1.2) or AddQueueMember (on 1.4) is the better solution to go with. Agentcallbacklogin has bugs and randomly crashes my system even though we probably don't have more than 100 callbacks a day. Unfortunately, the developers have decided that since you can get something similar using lots of dialplan logic, they have decided that it's easier to deprecate the function than to fix it. Psst -- don't tell the developers, but we could probably get something similar to Asterisk with a box of tin cans, a spool of string and a couple of carrier pigeons :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
Grandstream make a 4-port and an 8-port FXO ATA Linksys make one but I think it is proprietary to their own Linksys One system regards, Drew Mike wrote: Probably, if I only needed one FXO. What is the customer has 4 channels (PSTN lines)? Don't I need 4 FXO? And, about the Sipura, it looks like it would do what I want, but it only has one FXO, limiting it's usefulness. Mike On 4/10/07, *Mike* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone My own Asterisk - Outside lines But when it comes to smaller villages (I deal with people in tiny places), I'd like to reuse their own PSTN line this way: Customer's SIP hardphone My own Asterisk -- Some device on the customer's premise customer's PSTN lines I know ATAs are mostly used in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike Hello Mike, Wouldn't a Sipura SPA 3000, with an FXS and an FXO, handle what you want? Cheers, Francis -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
[EMAIL PROTECTED] wrote: On Tue, 10 Apr 2007 10:17:27 -0400 (EDT) Alex Balashov [EMAIL PROTECTED] wrote: And, sorry if I misunderstood, but you still have signaling, not signalling, in your zapata.conf? [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 10 07:35:44 WARNING[4960]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Apr 10 07:35:44 ERROR[4960]: chan_zap.c:7034 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 10 07:35:44 ERROR[4960]: chan_zap.c:10468 setup_zap: Unable to register channel '1' Apr 10 07:35:44 WARNING[4960]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Apr 10 07:35:44 WARNING[4960]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. Did you restart Asterisk after you made the changes? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble recording calls
On 4/9/07, ahester [EMAIL PROTECTED] wrote:
Hi all,
I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see got hangup request on the cli and
then mixmonitor ends.) I am guessing this has to do with the announce
file that is played to the engineer before the call is connected. It
seems that if the call rolls to the next queue because of timeout,
asterisk doesn't even try to record it. (I don't see any mixmonitor on
the cli for the next queue).
I would appreciate any help with this. I have to have all calls
recorded and I have to do announcements so that the callee knows how to
answer the phone.
Thanks,
Andy
The configs are as below:
From extensions.conf:
#after various menu stuff, send to support
exten = 214xxx,13,SetGlobalVar(ORIGIN=support)
exten = 214xxx,14,Queue(support1|tr|||10)
exten = 214xxx,15,Queue(support2|tr|||)
#dial command for sip extensions that are in the queues
exten =
_72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _72XXX,2,Dial(SIP/${EXTEN})
exten =
_73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _73XXX,2,Dial(SIP/${EXTEN})
queues from queues.conf:
[support1]
; Support call queue
announce = 16
strategy = rrmemory
timeout = 15
retry = none
wrapuptime=15
announce-frequency = 0
joinempty = no
leavewhenempty = yes
member = Agent/2008
member = Agent/2009
member = Agent/2014
member = Agent/2015
member = Agent/2017
member = Agent/2018
member = Agent/2019
member = Agent/3520
member = Agent/3521
member = Agent/3522
member = Agent/3524
member = Agent/3529
[support2]
; Support2 call queue
announce = 16
strategy = ringall
announce-frequency = 0
; Added below for testing because the second queue was not even trying
to record
; according to the asterisk console (still doesn't)
Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
monitor-format = wav
monitor-join = yes
joinempty = yes
member = SIP/72008
member = SIP/72009
You cannot use the MixMonitor app on its own in a callback scenario
because, as you've already discovered, MixMonitor senses the call
transition between the time the agent answers and the calls is then
bridged with the waiting caller and still stop recording.
To fix this, in the 1.4 version of app_queue, there's a
monitor-type=MixMonitor parameter which will use the MixMonitor
appropriately natively in app_queue instead of Monitor.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Verizon-Vonage Lawsuit
Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
Mike, No, not necessarily. There are plenty of off-the-shelf IP PBX units, and some of them are not expensive at all. I heard somewhere Linksys makes one for about $500 and it can amply serve a small office. Should be enough for your purposes. Either way, it'd be immensely cheaper than buying a PC, installing Linux, throwing Asterisk on it, setting it up manually, even if you manage to find a way to replicate and scale some of those processes. Unfortunately, I'm not a very good market researcher. :( -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
You can use a quad FXO gateway. Ask on the -biz list if you are looking for suppliers. Mike wrote: Thanks Alex, That was my original thought, to just buy a TDM400 from Digium and put in as many FXO as I wanted, but I liked having the ease of just buying something off the shelf, even if it meant paying a little more. But it looks like I won't have much of a choice. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Tuesday, April 10, 2007 10:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk Hi Mike, You should be looking at ATAs that have FXO, rather than FXS interfaces. Most ATAs come with FXS ports so that you can connect analogue phones to them, but in this case you're wanting to take PSTN lines from the outside, so FXO is desirable. Second, you'd have to make sure that the ATA supports the sort of application you're using it for; most are manufactured on the opposite premise. I am actually not sure offhand of any ATA firmware that I know that I imagine would work this way, although I'm confident it exists as consecutive back-to-back analogue-VoIP adaptations in many scenarios can get quite complex and requires that flexibility. Basically, you're looking for a small IP PBX that uses SIP internally among its private nodes and takes PSTN trunks from the outside. That's what PBXs typically do. :-) If all else fails, you can always roll your own functionality of this nature by using FXO cards in Asterisk. There are various distributions that package it in a very lightweight and reusable manner specifically for this type of purpose, or you can roll your own if it's scalable enough. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip_header=value?
Well here is my scenario,
Our users have option to register with one Primary did and 5 secondary dids
for the purpose of distinctive-ring/did-based-routing. If a user is
registered with us and he is using sipura, then we have to send 6 different
bellcores in alert into sip header for his different dids like this
Caller called user1 did1(123)-send bellcore1
Caller called user1 did2(456)-send bellcore2
and so on
If the user is using asterisk to register with us, then we also have to send
the dnid so that when the user receives the dialed num, he has to decide to
route that call to which extension based on dnid.
So the first problem:
For sending the dnid i used the dial application like this:
(1)--Dial(SIP/[EMAIL PROTECTED]) incase when the user is using asterisk as a
peer
(2)--Dial(SIP.user1) incase when the user is using sipura as a peer
i wanted to use (1) for dialing sipura also but it doesnt work, so i dial
like (1) if user is using asterisk and (2) for every other device. I am
still finding a way to solve this problem so that i dont have to check the
called user's useragent for every call. I was trying to find a way to send
the dnid in some header field. thats why i started this thread.
The second Problem:
Asterisk doesn't set the ${DNID} variable to the dialed extension extension
num. all of my system for setting the bellcore and sending the dnid is based
on ${DNID} variable, and i just came to know this problem. I dont know why
this i happening. if you know plz help.
So.any ideas
On 4/10/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
There is a difference bwtween the SPA-941 which is a standalone desktop
phone w/ Speakerphone, and your SPA2100
What header value are you trying to set, and why?
Because the example I gave is for making the SPA-941 an intercom, and also
requires some changes in it's settings within the phones web interface. So
of course my example isn't going to anything for your SPA-2100 except give
you an example of how to set a SIP header.
So now my question becomes, What is it that you are attempting to do by
setting the SIP Header?
Best Regards,
Karl J. Vesterling
Rizwan Hisham wrote:
I have discovered another thing,
exten= 123,1,Dial(SIP/abc/${EXTEN},,Tt)
exten= 123,2,hangup
If the user is using xlite to register with my asterisk server, then we
are able to call him using the above axtension, and if the user is using
sipura the the above extension does not dial the user instead it displays a
congestion message as before, maybe there is a problem in sipura firmware. I
am using Linksys/SPA2100- 3.3.6. any ideas why is sipra behaving like
this.
for sipura to ring we have to use the following extension, without
${EXTEN} variable
Dial(SIP/abc,,Tt)
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Apparently it sets a SIP_HEADER variable named Call-Info to a value of
answer-after=0 effectively telling the Sipura to answer the call and put
it through to speakerphone.
I will say that extensions.ael is a bit different from regular line
based extensions.conf in that I seem to have to escape all sorts of
stuff with the \ character that I don't have to in extensions.conf
Back to work, I'll check in on this thread later this evening.
Rizwan Hisham wrote:
I dont understand it
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
whats it doing here?
On 4/9/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael
Rizwan Hisham wrote:
I have tried it, it doesnt work
On 4/9/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
is there anyway i can set SIP_HEADER(To) to the value i like?
If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Regards
Rizwan Hisham
Software Engineer
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OT: Re: [asterisk-users] maximum simultaneous calls
Steve Edwards wrote: *snipped Psst -- don't tell the developers, but we could probably get something similar to Asterisk with a box of tin cans, a spool of string and a couple of carrier pigeons :) don't forget the sneakers! G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 38
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] maximum simultaneous calls
James FitzGibbon wrote: Were your problems limited to queue weights, or are agent penalties likely to cause this as well? I'm about to embark down the ACD path, and still unsure as to whether AgentCallbackLogin (on 1.2) or AddQueueMember (on 1.4) is the better solution to go with. Queue member penalties (there are no 'agent' penalties as agents are concept provided by chan_agent and it does not handle penalties) are not known to cause any problems. Weights (between queues) are definitely problematic, primarily because they cause a great deal of extra locking/unlocking operations as all the queues and their callers are traversed trying to make a decision which member should get which call. The 'autofill' behavior that has been recently added only makes this worse. We plan on rebuilding the queuing system from scratch, using a state-driven rather than polling model, which will pretty much eliminate this issue as there will be a lot less 'work' involved in delivering calls to queue members. Most of the design work for that will happen at next month's DevCon, and then you will see the code begin to appear in the following weeks. It is possible that the code will be backportable to Asterisk 1.4, although obviously we won't be doing that as part of any official releases. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum retries exceeded on transmission
Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
On Tue, Apr 10, 2007 at 07:37:38AM -0700, [EMAIL PROTECTED] wrote: On Tue, 10 Apr 2007 10:17:27 -0400 (EDT) Alex Balashov [EMAIL PROTECTED] wrote: And, sorry if I misunderstood, but you still have signaling, not signalling, in your zapata.conf? [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 10 07:35:44 WARNING[4960]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Apr 10 07:35:44 ERROR[4960]: chan_zap.c:7034 mkintf: Unable to open channel 1: No such device or address No device to handle /dev/zap/1 ? I guess that the module is not loaded . here = 0, tmp-channel = 1, channel = 1 Apr 10 07:35:44 ERROR[4960]: chan_zap.c:10468 setup_zap: Unable to register channel '1' Apr 10 07:35:44 WARNING[4960]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Apr 10 07:35:44 WARNING[4960]: loader.c:554 load_modules: Loading module chan_zap.so failed! What is the output of: cat /proc/zaptel/* lsmod | grep zaptel ls -l /dev/zap -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
On Wed, Apr 04, 2007 at 05:52:46PM +0100, Chris Blunt wrote: Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! maybe ztdummy.ko was not regenerated? 'make clean' is normally not needed when changing kernel versions, as Kbuild is usually smart enough to tell the difference. What is the output of: modinfo ./ztdummy.ko -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
On Tue, Apr 10, 2007 at 10:58:45AM -0300, Alejandro Cabrera Obed wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? http://packages.debian.org/asterisk (Hey, Etch is out! oldstable no longer has Asterisk 0.1 ;-) As you can see, there is 1.2.16 (and soon 1.2.17, I've already asked to upload it) in Sid, and 1.4.2 in Experimental . Etch has 1.2.13 . Alternatively, try: deb http://updates.xorcom.com/rapid etch main which has some backports of Sid packages. 2) Do I have to configure a dummy PSTN interface in my case ?? You need the zaptel module ztdummy. As you just need ztdummy and not a real zaptel, there's really no reason to use latestgreatest bleeding-edge zaptel. If you added my packages source from above: apt-get install zaptel zaptel-modules-`uname -r` /etc/init.d/zaptel start If you have just the standard Etch sources, the procedure is a bit more complicated, because you have to generate the package zaptel-modules for your kernel: apt-get install zaptel zaptel-source build-essential # maybe you need to also explicitly install linux-headers-`uname -r` # to build and install the zaptel-modules package for your kernel: # (Will probably fetch the proper linux-headers package as well) m-a a-i zaptel /etc/init.d/zaptel start In both cases You should get an error from ztcfg because there's no zaptel.conf, but just ignore it, as you don't need ztcfg for ztdummy. To make that error disappear you can run: touch /etc/zaptel.conf And if you have a debian-asterisk howto, I really thank you. As usual with Debian, start from /usr/share/doc/PACKAGE/README.Debian . Two other potentially-useuful packages in our repository: freepbx # though still a bit broken. maybe try # 'freepbx-common freepbx-modules' asterisk-config-simple Maybe they'll also help you getting started. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding chan_celliax
Giovanni Maruzzelli wrote: I was unaware of that build, thanks Tzafrir! But it seems old... You can download the current complete sources with: svn checkout http://www.celliax.org:8081/svn/celliax/branches/test1 test1 or you can download a livecd from www.celliax.org and test it without install anything. For compiling yourself you actually just need the files that build chan_celliax, that you find in test1/celliax_stuff/ (chan_celliax.c chan_celliax_spandsp.c chan_celliax_spandsp.h) and to modify the channels/Makefile And, of course, the configuration files (particularly celliax.conf) that you find in celliax_stuff/newconfigs/*. So, if you would better like to compile with the asterisk-dev packages, just download the svn sources as told, put those three files in the asterisk/channels directory and modify the asterisk/channels/Makefile. Anyway, if you download and build directly from the svn sources (just make install from the test1/asterisk-1.2.17 directory), it will put all the stuff in /usr/local/asterisk, /usr/local/asterisk/etc/asterisk/*.conf, /usr/local/asterisk/usr/sbin/*, etc, so it will not clutter your computer and other existing asterisk installations (you will have to remove just the /usr/local/asterisk directory). The test1 branch of the svn is the latest and greatest, but do not yet support skype and alsavoicemodems. The trunk of the svn supports skype and alsavoicemodems, but... ;-) Giovanni Hi, thanks a lot for your directions, I've downloaded the svn from celliax, but i wasn't aware than these sources are the full asterisk sources including asterisk and chan_celliax. So, i thinks than it's a good idea to compile the sources, build a deb and forget about the debian official deb. Is that right or there is a more debian way to add this channel to the debian release. thanks a lot -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com Spelling is a lossed art. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] XO Flex T-1 Asterisk
I have an XO t-1 line which includes VOIP on 5 lines. When a call comes in, it drops the bandwidth on the t1 and when the call is over bandwidth is restored. The provided a channel bank which does this. My question is, if I use asterisk, then am I losing double bandwidth for each call? For example the only way I guess I can connect to the lines is from the 66block they provided which would serve as POTS lines. So, I guess I need to connect them to asterisk using FXO cards? My extension would take bandwidth from the t1 and the actual call running over XO would also take bandwidth, right? If someone could explain the best way for me to set this up, I would really appreciate it. Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
Tzafrir Cohen wrote: On Tue, Apr 10, 2007 at 10:58:45AM -0300, Alejandro Cabrera Obed wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? http://packages.debian.org/asterisk (Hey, Etch is out! oldstable no longer has Asterisk 0.1 ;-) As you can see, there is 1.2.16 (and soon 1.2.17, I've already asked to upload it) in Sid, and 1.4.2 in Experimental . Etch has 1.2.13 . Alternatively, try: deb http://updates.xorcom.com/rapid etch main which has some backports of Sid packages. 2) Do I have to configure a dummy PSTN interface in my case ?? You need the zaptel module ztdummy. As you just need ztdummy and not a real zaptel, there's really no reason to use latestgreatest bleeding-edge zaptel. If you added my packages source from above: apt-get install zaptel zaptel-modules-`uname -r` /etc/init.d/zaptel start If you have just the standard Etch sources, the procedure is a bit more complicated, because you have to generate the package zaptel-modules for your kernel: apt-get install zaptel zaptel-source build-essential # maybe you need to also explicitly install linux-headers-`uname -r` # to build and install the zaptel-modules package for your kernel: # (Will probably fetch the proper linux-headers package as well) m-a a-i zaptel /etc/init.d/zaptel start In both cases You should get an error from ztcfg because there's no zaptel.conf, but just ignore it, as you don't need ztcfg for ztdummy. To make that error disappear you can run: touch /etc/zaptel.conf And if you have a debian-asterisk howto, I really thank you. As usual with Debian, start from /usr/share/doc/PACKAGE/README.Debian . Two other potentially-useuful packages in our repository: freepbx # though still a bit broken. maybe try # 'freepbx-common freepbx-modules' asterisk-config-simple Maybe they'll also help you getting started. Dear people, thanks for your help...I appreciatte it a lot. But one more question please: I have a Debian host base with vserver support (virtual machines, I use them for running squid, postfix and a lot of services without problems) I?ve just installed Asterisk in a new vserver from Debian Etch repositories and I get this error: Setting up zaptel (1.2.11.dfsg-1) ... mknod: `/dev/zap/ctl': Operation not permitted dpkg: error processing zaptel (--configure): subprocess post-installation script returned error exit status 1 After that I see the content of /dev/zap and there is nothing at all. Any idea ??? Can I continue without this device if I use only softphones ??? Thanks again Alejandro -- Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificacio'n Nacional Tributario y Social Consejo Nacional de Coordinacio'n de Poli'ticas Sociales Presidencia de la Nacio'n Julio A. Roca 782 - Piso 5 Ciudad Auto'noma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 [EMAIL PROTECTED] NOTA DE RESPONSABILIDAD: -- Este mensaje proviene de Internet,tome los recaudos necesarios en su manejo. El contenido del presente mensaje y sus adjuntos es privado, estrictamente confidencial y exclusivo para su destinatario, pudiendo contener informacio'n protegida por normas legales y de secreto profesional. Bajo ninguna circunstancia su contenido puede ser transmitido o revelado a terceros ni divulgado en forma alguna. En consecuencia de haberlo recibido solicitamos contactar al remitente y eliminarlo de su sistema. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] XO Flex T-1 Asterisk
'Your extension' would only use the bandwidth if it is off-site. If your
phone ('extension') is on the LAN then it 'should' not touch the T1.
Furthermore, the XO product is not compatible with Asterisk unless you
do as you say and connect the FXO or T1 port to your asterisk server.
You will still need to use XO's IAD. (Integrated Access Device)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of mypaisa
Sent: Tuesday, April 10, 2007 2:00 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] XO Flex T-1 Asterisk
I have an XO t-1 line which includes VOIP on 5 lines.
When a
call
comes in, it drops the bandwidth on the t1 and when the call is over
bandwidth is restored. The provided a channel bank which does this.
My question is, if I use asterisk, then am I losing double bandwidth
for each call? For example the only way I guess I can connect to the
lines is from the 66block they provided which would serve as POTS
lines. So, I guess I need to connect them to asterisk using FXO
cards? My extension would take bandwidth from the t1 and the actual
call running over XO would also take bandwidth, right?
If someone could explain the best way for me to set this up, I would
really appreciate it.
__
__
Now that's room service! Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] ISDN BRITE support?
Hi all, I found some message in the Digium list archives that discussed ISDN BRITE support. There was also some discussion on 4:1 bitrate conversion. I did some searching on the source code and didn't see any reference to BRITE or bitrate conversion. Does the current code properly pass a D-channel from a BRI port to a DS0 channel on a channelized T1? BRITE seems pretty straight forward, but I didn't see any examples on how to do this. Are there any plans for supporting the more complicated bitrate conversion that muxes the D-channels from BRI ports to a single DS0 channel? The context for these questions is the same as what was previously discussed. I would like to use Asterisk to route H.320 video conferencing equipment into a T1 and provide a platform for eventually moving off our old PBX. ...Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf
On Tue, 10 Apr 2007 19:53:01 +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Apr 10, 2007 at 07:37:38AM -0700, [EMAIL PROTECTED] wrote: On Tue, 10 Apr 2007 10:17:27 -0400 (EDT) Alex Balashov [EMAIL PROTECTED] wrote: And, sorry if I misunderstood, but you still have signaling, not signalling, in your zapata.conf? [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 10 07:35:44 WARNING[4960]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Apr 10 07:35:44 ERROR[4960]: chan_zap.c:7034 mkintf: Unable to open channel 1: No such device or address No device to handle /dev/zap/1 ? I guess that the module is not loaded . Solved, you are right. Once the module is loaded now it works. Thanks -- Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Too much silence, perceived delay
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:51 PM: Joe Acquisto wrote: Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM: Have you been able to test this yourself? (Three to four seconds seems inordinately long. That's as bad as a satellite link.) No, not tested by me, I only heard about it today, via email. I don't doubt that they are noticing some delay, I just question how extreme it is. Have you tried tinkering with the gain settings? Adjusting the gain can impact sidetone, which might improve the call experience. No, not yet. Any suggestions as to direction and magnitude? After confirming that they're experiencing what they say they've been experiencing, I would start with the rxgain and increment it by 2 or 3, then test. Applying rxgain of 2 seems to have satisfied the user who was complaining. My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial from SCCP - SIP phone with no problems, but not SIP - SCCP or SCCP - SCCP. I have a feeling I'm forgetting something fairly easy and stupid, but I can't seem to see what it is. Anyone have any suggestions? sccp.conf [general] keepalive = 30 context = internal bindaddr = 192.168.1.1 port = 2000 debug = 10 firstdigittimeout = 16 digittimeout = 8 [devices] type= 7960 description = Cisco1 tzoffset= 0 autologin = 104 ; speeddial = 101, 105 device = SEP00036BC3852B [lines] id= Cisco1 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 500 vmnum = 500 cid_name = Cisco1 cid_num = 104 line = 104 extensions.conf [internal] include = outbound-local include = outbound-long-distance ; Software phone exten = 101,1,Dial(SIP/test-softphone,,r) exten = 102,1,Dial(SIP/bob,20) exten = 102,2,Voicemail(u102) exten = 102,102,Voicemail(b102) exten = 102,103,Hangup() exten = 103,1,Dial(SIP/bill,20) exten = 103,2,Voicemail(u103) exten = 103,102,Voicemail(b103) exten = 103,103,Hangup() exten = 104,Dial(SCCP/SEP00036BC3852B,20) exten = 104,2,Voicemail(u104) exten = 104,102,Voicemail(b104) exten = 104,103,Hangup() exten = 105,Dial(SCCP/SEP00036B095612,20) exten = 105,2,Voicemail(u105) exten = 105,102,Voicemail(b105) exten = 105,103,Hangup() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Too much silence, perceived delay
On Tue, 10 Apr 2007, Joe Acquisto wrote: My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. Anytime somebody complains of delay or lag, have them call a cell phone from a cell phone and listen to themselves. Usually their jaw drops :) Then I ask them when was the last time somebody asked them to call back on a land line because the delay was too long. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: exten = 104,Dial(SCCP/SEP00036BC3852B,20) exten = 104,2,Voicemail(u104) exten = 104,102,Voicemail(b104) exten = 104,103,Hangup() Off the top of my head, I would say that your dial statement should be Dial(SCCP/104,20). You should be dialing the line, not the device. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Too much silence, perceived delay
Steve Edwards wrote: On Tue, 10 Apr 2007, Joe Acquisto wrote: My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. Anytime somebody complains of delay or lag, have them call a cell phone from a cell phone and listen to themselves. Usually their jaw drops :) Then I ask them when was the last time somebody asked them to call back on a land line because the delay was too long. There are different types of delay. There is audio delay and there is dialing delay. I suspect the users are complaining about dialing delays, rather than audio delays. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T100P -- TE120P
On 4/9/07, Carlos Chavez [EMAIL PROTECTED] wrote: No, that particular model is not able to use an E1. You need a TE110P which is able to select between both. They used to have an E100P card that was E1 only, both were replaced by the TE110P. The TE120P is the updated version of the TE110P and is much more compatible than the TE110P. It also has a hardware echocan port (the TE110 doesn't). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On Tue, Apr 10, 2007 at 02:51:31PM -0500, Lacy Moore - Aspendora wrote: On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: exten = 104,Dial(SCCP/SEP00036BC3852B,20) exten = 104,2,Voicemail(u104) exten = 104,102,Voicemail(b104) exten = 104,103,Hangup() Off the top of my head, I would say that your dial statement should be Dial(SCCP/104,20). You should be dialing the line, not the device. Tried that as well. -Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
- [EMAIL PROTECTED] wrote: [snip] I have a feeling I'm forgetting something fairly easy and stupid, but I can't seem to see what it is. Anyone have any suggestions? Dial(SCCP/[EMAIL PROTECTED]) -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Too much silence, perceived delay
Eric ManxPower Wieling [EMAIL PROTECTED] Wrote: 4/10/2007 3:53 PM: Steve Edwards wrote: On Tue, 10 Apr 2007, Joe Acquisto wrote: My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. Anytime somebody complains of delay or lag, have them call a cell phone from a cell phone and listen to themselves. Usually their jaw drops :) Then I ask them when was the last time somebody asked them to call back on a land line because the delay was too long. There are different types of delay. There is audio delay and there is dialing delay. I suspect the users are complaining about dialing delays, rather than audio delays. During conversation about this, both types of delay were mentioned. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
Do you have any console messages? SCCP uses a station start tone message with a value of Inside Dial Tone and a direction of Tone Output User. and the line instance and Station Tone Output Direction should be set to something other than 0. SCCP runs over TCP so you should get this message, but it would be interesting to see if you get this message the phone still doesnt play dial tone. My experiences with chan_sccp have been disappointing at best. If you can get a trace of both SCCP legs send it to me and i can take a look at it. -- Original message -- From: Jason Parker [EMAIL PROTECTED] - [EMAIL PROTECTED] wrote: [snip] I have a feeling I'm forgetting something fairly easy and stupid, but I can't seem to see what it is. Anyone have any suggestions? Dial(SCCP/[EMAIL PROTECTED]) -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
can't seem to see what it is. Anyone have any suggestions?
Dial(SCCP/[EMAIL PROTECTED])
--
That seems better, but it's almost as if asterisk doesn't realize that the
phone has been hung up.
[Apr 10 16:27:02] ERROR[31001]: sccp_channel.c:531 sccp_channel_hold:
SEP00036BC3852B can't put on hold an inactive channel 104-1
-- SEP00036BC3852B: Display prompt on line 1, callid 1, timeout 5
-- SEP00036BC3852B: Got message StimulusMessage
-- SEP00036BC3852B: Got stimulus=VoiceMail (15) for instance=1
-- SEP00036BC3852B: Voicemail Button pressed on line (1)
-- SEP00036BC3852B: Getting the active channel on device
-- SEP00036BC3852B: Sending digits 500
-- SEP00036BC3852B: Sending digit 5
-- SEP00036BC3852B: Sending digit 0
-- SEP00036BC3852B: Sending digit 0
-- SEP00036BC3852B: Got message OffHookMessage
-- SEP00036BC3852B: Getting the active channel on device
-- SEP00036BC3852B: Taken Offhook with a call (1) in progess. Skip it!
sterisk*CLI sccp show lines
asterisk*CLI
NAME DEVICE MWI Chs Active Channel
=
105 SEP00036B095612 OFF 0--
104 SEP00036BC3852B OFF 1InvalidNumber Outbound 105
0x8 (alaw)
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Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: exten = 104,Dial(SCCP/SEP00036BC3852B,20) exten = 104,2,Voicemail(u104) exten = 104,102,Voicemail(b104) exten = 104,103,Hangup() Actually, if this is a cut and paste, you are missing the 1. It should be: exten = 104,1,Dial... you have exten = 104,Dial... Also, jumping to n+101 is not the default in Asterisk 1.2+, you might want to search the wiki (www.voip-info.org) for priority jumping and it can explain much better than I can. A better question, I guess, is this chan_sccp or chan_skinny? If chan_sccp did you successfully compile it with the patches for 1.4? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
So, a packet trace, at router-internet, was done. Not much to speak of, filtering for phone/* server traffic. While I can see what appears to be a session initiation and they make nice, there appears to be no traffic for audio, at all. Anyone have an example they could share? Or is someone quite well versed in SIP traffic who can read the trace? joe a. Joe Acquisto [EMAIL PROTECTED] Wrote: 4/9/2007 1:42 PM: Hi. Is there a way to isolate what shows on CLI to just the conversation with that extension? There appears to be a lot of stuff unrelated to this extension. Packet traces are not out of the question, but cannot be done today. joe a. Yossi Ben Hagai [EMAIL PROTECTED] Wrote: 4/9/2007 12:56 PM: Hi Joe, The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call. You can however tell that something is wrong since the message is being retransmitted since the server didn't receive 200 OK in reply - while it could be due to the client being offline or not supporting this feature It could imply a NAT issue so try to recheck your NAT configs. can you post a full trace (starting with the INVITE message)? also you can try to run a sniffer trace on the client side to see if it receives/sends the messages correctly. Joss. On 4/9/07, Joe Acquisto [EMAIL PROTECTED] wrote: I never get this far, apparently. While the connection seems to be made, and calls can be completed (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: nsip:[EMAIL PROTECTED];tag=as67e5c857 To: nsip:[EMAIL PROTECTED];tag=9c58a77e Contact: sip:[EMAIL PROTECTED] Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 - Does this imply anyting to anyone? Call can be made, after this. joe a. ** dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM: joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office- - --|sip phone |-ditto Hope that is intelligible. joe a ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] adding chan_celliax
On Mon, Apr 09, 2007 at 04:41:42PM -0500, Patricio Valarezo Lozano wrote: Hi, thanks a lot for your directions, I've downloaded the svn from celliax, but i wasn't aware than these sources are the full asterisk sources including asterisk and chan_celliax. So, i thinks than it's a good idea to compile the sources, build a deb and forget about the debian official deb. Is that right or there is a more debian way to add this channel to the debian release. Debian already has several packages of out-of-tree asterik modules, that are built vs. asterisk-devel: * asterisk-addons * The spandsp modules (fax, dtmftotext) * chan_capi-cm This is how I built my deb. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
On Tue, Apr 10, 2007 at 03:10:32PM -0300, Alejandro Cabrera Obed wrote: Dear people, thanks for your help...I appreciatte it a lot. But one more question please: I have a Debian host base with vserver support (virtual machines, I use them for running squid, postfix and a lot of services without problems) I?ve just installed Asterisk in a new vserver from Debian Etch repositories and I get this error: Setting up zaptel (1.2.11.dfsg-1) ... mknod: `/dev/zap/ctl': Operation not permitted dpkg: error processing zaptel (--configure): subprocess post-installation script returned error exit status 1 Right. http://bugs.debian.org/411850 Workaround: edit /var/lib/dpkg/info/zaptel.postinst to remove the mknod calls and run 'apt-get install -f' You'll have to find a way to generate the zaptel device files. Maybe ask the vserver host maintainer. The device files you need: /dev/zap/ctl /dev/zap/pseudo Is /dev/zap/channel also needed? After that I see the content of /dev/zap and there is nothing at all. Any idea ??? Can I continue without this device if I use only softphones ??? Yes, but you won't have zaptel timing (e.g: no meetme) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How well does a celldock work with Asterisk?
to all, I have a cell interface that hooks up to a standard pots handset... can use a cingular, tmobile, or SIM card provider.. hookup is about 30 seconds has a remote antenna so you can locate the unit about 10 ft from the antenna... quality is good... well, as good as cingular anyway... :) contact me off list for more info. daveC Stephen Bosch wrote: Joe Acquisto wrote: Sometimes it's just a matter of finding a clean pair in the cable. Have you tried asking Verizon to fix the problem? Don't get me started. That's how I know so much about the situation. They seem disinclined to address the matter, except with happy talk about FIOS in my future. Soon. Right after the metro areas are done. Right. The only fiber around here will be in my diet. Hehe. Are you in a rural area? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels
vijay, I had a similar problem with a pots line and 1.4.1... zap wasn't loading. from the CLI check that zap is loaded with 'zap show channels' pbx15*CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 25defaultdefault 26defaultdefault 27defaultdefault 28defaultdefault pbx15*CLI it this is ok, then check the kernel zap driver with [EMAIL PROTECTED] asterisk]# lsmod | grep zap zaptel190244 27 zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 1920 2 zaptel,hisax [EMAIL PROTECTED] asterisk]# to see if the zap driver is loaded. then check /etc/zaptel.conf # # We are all done with our channel parameters, so now we specify what # # channels they apply to # channels=1-4 fxoks=25,26 fxsks=27,28 tail the file and you should see some fxo/fxs config entries... if you install * w/o the board installed, in some cases, I have seen * forget to load the zaptel drivers, or perhaps that was trixbox or asteriskNow, I forget... daveC Vijay Gaur wrote: Yes when I plug my phone to vonage adapter it rings fine. I will run and send you the output soon. Thanks Vijay On 4/9/07, *Stephen Bosch* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Vijay Gaur wrote: I hear ring few times and then it goes to voice mail. Looks like call is not going to asterisk. My regular phone attached to that line works fine. When you plug the phone into the port on the Vonage ATA that you're using to connect to Asterisk, the phone rings when you call the number? Here's my next question: What does cat /proc/interrupts show on the Asterisk server? Run that and post the output. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote SIP, no audio, or one way audio.
joe, look for the codec negociation... I have a similar problem where the endpoints could not agree on the codec and thus no audio went through. in 1.4.X CLI sip set debug peer extension yields, Audio is at 10.10.15.15 port 15342 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (no NAT) to 10.10.15.219:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 make sure both endpoints have at least one codec that is the same... if not, adjust your sip.conf for both endpoints. daveC Joe Acquisto wrote: Hi. Is there a way to isolate what shows on CLI to just the conversation with that extension? There appears to be a lot of stuff unrelated to this extension. Packet traces are not out of the question, but cannot be done today. joe a. Yossi Ben Hagai [EMAIL PROTECTED] Wrote: 4/9/2007 12:56 PM: Hi Joe, The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call. You can however tell that something is wrong since the message is being retransmitted since the server didn't receive 200 OK in reply - while it could be due to the client being offline or not supporting this feature It could imply a NAT issue so try to recheck your NAT configs. can you post a full trace (starting with the INVITE message)? also you can try to run a sniffer trace on the client side to see if it receives/sends the messages correctly. Joss. On 4/9/07, Joe Acquisto [EMAIL PROTECTED] wrote: I never get this far, apparently. While the connection seems to be made, and calls can be completed (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: nsip:[EMAIL PROTECTED];tag=as67e5c857 To: nsip:[EMAIL PROTECTED];tag=9c58a77e Contact: sip:[EMAIL PROTECTED] Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 - Does this imply anyting to anyone? Call can be made, after this. joe a. ** dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM: joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office-- --|sip phone |-ditto Hope that is intelligible. joe a ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and
Re: [asterisk-users] intermittent choppy sound over wifi link
On Sun, 8 Apr 2007, Curt Shaffer wrote: I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider - IAX2 over the Internet - 20Mb fiber connection - router - Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running 3.1.19(SE) firmware), also tested with X-lite softest - PIX 506 (although I have tried multiple routers and direct connection to the radio try to fix the problem) - 1 mile 802.11b link to AP - 15 mile 802.11b link Backhaul - router - Asterisk I'm, er, impressed. Some years ago I was involved with community broadband networks delivered via Wi-Fi, and the results were dissapointing to say the least. I was using smartbridges kit - good kit at the time and designed for the great outdoors. Relatively expensive though. One thing I found was putting 2 Wi-Fi links back to back - ie. with just a switch in-between, would seriously degrade bandwidth through the link. And 15 miles! That's really stretching it, but I guess you have the kit - big parabolics or dishes, line of sight, no fresnel zone intrusions, etc. We tried daisy-chaining 4 links together and struggled to get 1Mb/sec through it. What kills Wi-Fi is full duplex. It's only half-duplex kit, so there is a turn-around on the link to simulate full duplex. When your packets are short and coming in both directions, the radio turn-around time can exceed the packet time. Put 2 links back to back through a switch or hub and it's worse. Put short packets through it at regular intervals and it's worse yet. Try to run short packets both ways at the same time (which you have to for VoIP) and it's even worser. But you'll get good ping times and downloading data will appear just fine which is what'll make it all the more frustrating! So make sure there is nothing else on the Wi-Fi links, especially no uploads. My Asterisk version is Asterisk 1.2.12.1, Zaptel 1.2.9.1. Ping times are ~10ms, jitter is under 10 with an average of 5. QoS is enabled in the router for SIP, RTP and IAX2 traffic going to and from the Asterisk box. QoS won't help you at the end of 2 radio hops, because by the time the packet gets to the router, it's gone over 2 Wi-Fi hops and it's too late for the router to do anything with it, so uploads are to be avoided. That's what used to kill my networks (I had 3 community networks with a few 100 people in total, connected togther with a fibre backhaul) One kiddie using p2p software would kill his entire segment of the network as the uploads would cripple any other downloads, and because I only had one router per community, he'd effectively cripple everyone in that community until I cut him off and/or employed some really harsh traffic shaping which wan't that effective anyway, but helped. I lusted after a router at each AP, but that was never going to happen as we never had the money... When I experience the choppiness the ATA reports packet loss on the web interface (Call 1 Packets Lost: ). I can run something such as ping plotter from the same leg of the network that the Asterisk box is on while this is happening and there is not even a small glitch of lost packets on the network but the ATA displays otherwise. The only thing I have come up with thus far is possible retransmissions on the wireless connection (and due to the type of gear, I'm not able to see this data). We are way out in the country with no other real providers even close so I'm doubting interference although I suppose it is a possibility keeping an open mind. My question is can anyone point me to any possible reasons this would be happening? Also can anyone tell me other reasons other than real lost packets that the ATA would show this? My only guess on that was packets that never got an ACK due to server congestion or some other reason other than actual loss. As far as I'm aware, there's no ACK in the RTP stream - it's just a UDP stream with the bare minimal of overhead to help sequence and time packets. You may find that the Wi-Fi gear is dropping UDP packets if it gets overloaded. Can you increase the sample time - from (eg) 20ms to 40ms? So the packets are bigger and the radio turn-arounds are less frequent? Have you tried a lighter codec? (eg GSM - smaller packets, but less often?) Other than that, all I can suggest is Good Luck ... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On Tue, Apr 10, 2007 at 03:35:23PM -0500, Lacy Moore - Aspendora wrote: On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: exten = 104,Dial(SCCP/SEP00036BC3852B,20) exten = 104,2,Voicemail(u104) exten = 104,102,Voicemail(b104) exten = 104,103,Hangup() Actually, if this is a cut and paste, you are missing the 1. It should be: exten = 104,1,Dial... you have exten = 104,Dial... Sorry, it was a typo Also, jumping to n+101 is not the default in Asterisk 1.2+, you might want to search the wiki (www.voip-info.org) for priority jumping and it can explain much better than I can. A better question, I guess, is this chan_sccp or chan_skinny? If chan_sccp did you successfully compile it with the patches for 1.4? chan_sccp with the patches for 1.4 Thanks Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I co-invented that one. It's a good one. A lot of my input went into it, but the final product was much more general than I was originally led to believe. Somehow this patent was slowly changed from disclosing sip contact center technologies to a patent on using SIP. The original intention was to do this for disclosure purposes in order defend against clowns like Katz. However, the company that owns this patent has since transferred rights to one of their subsidiary IP PBX firms and eventually they may decide to use this patent for other purposes besides defensive disclosure. I imagine that they could always whip this patent out on competing SIP PBX companies... It certainly would be annoying to deal with. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, April 10, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemming=on It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, April 10, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: chan_sccp with the patches for 1.4 Everything should be fine, then, unless maybe you have really old firmware on the phones. That's the only thing I can think of. I've been running 1.4.2 with chan_sccp for a while in a test environment and have noticed no issues on 7960s and 7910s. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to install asterisk on redhat ?
Hiasterisk users... how to install asterisk on redhat ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to install asterisk on redhat ?
On Wed, 11 Apr 2007, Malik Mulki (Plant, Feed, Makassar) said something to...: Hiasterisk users... how to install asterisk on redhat ? There are numerous installation guides on this subject. But in general, you can either install a contributed RPM, or download the source code and compile it (along with libpri and zapata telephony interface if you need them). Check out ftp://ftp.digium.com/ -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] help with Sipura SPA 3000
I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. Get the impedance settings right. An impedance mismatch will cause echo (but may not be the only cause) But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. I think the 'echo suppression' setting causes this. It is meant to reduce the incoming audio (and hence the echo) while you are talking, which can be annoying but is supposed to be less annoying than the echo itself. 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! A dialplan of '(S0:s)' will get your phone to jump straight into the 's' extension in asterisk as soon as someone picks it up. From there you can do something like: [sip_ata_incoming] exten = s,1,Answer exten = s,n,DISA(no-password|sip_extension_in) so Asterisk will give you dialtone and do the dialplan stuff for you. From the 'sip_extension_in' context you can make a single '0' or '*' call the PSTN line. Good luck with the echo situation. I have an spa3000 and no matter what I do I get echo coming back to me with almost no reduction in volume!!! James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Learn some terminalogy before mounting this task.
All, I have done research on VoIP for some time now. I'm a Cisco certified Network Engineer however Telecom is not my strongest suit. I've been a part of this mailing list for sometime but my delusions of grandeur in migrating our 25 year old phone system to a new platform have been on the back burner, until now. I have found my company is moving to a new location(building) and this provides perfect opportunity. Long story short, I'm very Linux savvy having no problems compiling, building, making etc.. However getting connected to the PSTN is puzzling me. My vocabulary is lacking and I need to call our provider this week and get some circuits moved. So, my confusion (Current Setup) We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Then a cable from the T card(SmartCard) to the channel bank. From the Channel bank lines are punched down to the block. Then those ANALOG lines are fed into our Big hunking PBX mounted on the wall and two (Looks to be Rj11) lines come from it into our VM server. QA.. We are going to leave the telecom hardware behind.. I want to replace it all with an Asterisk or Tribox solution. I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? Does a channel (In terms of the T1) = a port? How many phone calls can one TDM400 support concurrently? (four ??) Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? Or Have a channel bank installed and use an OpenSwitch12 solution working with the analog signal? If we go with a Zapata T1 card for the Asterisk server would we be able to provision an analog phone line, for say a FAX machine from it? (Further information) Ultimately this is the FIRST of 3 offices that will be migrated. This first office has 30 physical phones and currently no DID (Hoping to change that as well) Once the second and third offices migrate(about the same number of stations) we would connect the 3 Asterisk systems over the WAN giving us free long distance between offices. The toughest challenge I foresee is getting overhead intercom/paging to work(We must be able to hail our warehouse staff over head), also Music on Hold... If I use Cisco 7940's (SIP) (Or the like) I'd like to integrate our phone extensions for a 3 offices into the directory Does this sound do-able? -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mounting this task.
Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices, too, especially if you go shopping with some friendly CLECs. The rule of thumb in the industry is that generally, once you pass the threshold of six or seven POTS lines, it becomes economical to just order an entire PRI, and once you do that, there usually aren't *very* considerable savings to be gained from turning down all but a few channels. A PRI has 23 channels (bearer channels (B channels)) and one signaling channel (D channel); it's a type of T1-based ISDN interface. So, you might potentially be able to get 23 in/outbound phone lines for roughly the same cost or a modest increase, which would increase your organisation's capacity to do things like conference calling and other things which tie up large amounts of outside lines. Do beware that if you go this route, PRIs can be ordered as inward-only (typically used for modem and termination-only telephony applications like voicemail, IVR, conferencing, etc.) or bidirectionally. If we go with a Zapata T1 card for the Asterisk server would we be able to provision an analog phone line, for say a FAX machine from it? No, not if the card doesn't have FXS ports on it. But you could get another Digium or Digiumlike card that does, even if it's just a single-port (like the hugely popular X100P, which is very inexpensive) and pull that off. Let me know what else we can answer, or if I substantially misunderstood your question. Good luck, -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mounting this task.
On Tue, 10 Apr 2007, Alex Balashov said something to this effect: Personally, the easiest approach I would say would be to order a PRI. The reason I say that, by the way, is because most T1-based digital voice trunking is done using PRIs these days. That doesn't mean you couldn't get a good old-fashioned EM trunk -- which is what would typically be used on a classic channelised T1 (which uses channel-associated signaling (CAS) rather than out-of-band PRI signaling on the non-bearing D channel), sometimes at a considerable discount from the telco. The Zaptel-compatible T1 cards support that as well. But PRI is a lot easier to manage for a variety of reasons IMHO, and is generally preferred. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Thanks for the answer. I've never heard that one before. I remember once I used a 3rd party component set (Indy 9) to do some smtp alert emails from a Windows application. Couldn't get the mails to go through for a few particular customers and after some research and talking to ISP's, we found out that the component set that I used was used to write spamming software and the headers produced by the component set were flagged by spamm assasin and others. Arg. Mail admins usually treat non-MTA software that send email as highly potential sources of spam. In your case, a whitelisting arrangement would take care of the problem or whatever arrangement they may have. Properly managed ISP's/domains will have a reachable contact at [EMAIL PROTECTED]/domain. The third party component set you mentioned...is it the original of indyproject.org too? The Indy 9 component most probably got itself into blacklists because of the IntraWeb framework, which most probably uses Indy 9 for smtp, has inadequate measures against abuse and the Indy 9 component creates an easily identifiable header. Whole server farms running proper MTA software will get blacklisted if they have abusable scripts or other stuff using them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Learn some terminalogy before mounting this task.
Alex, First may I offer thanks for your comment and reply. Most I understood and all I appreciate. To clarify the existence of the Channel bank at our current premise: In past years the Data and Voice WAS on the same T1. The 24 channels were used(The T1 was full) and the channel bank separated (At that time marketed as-) DPL (Digital Private Line) circuits to our two branch offices. That has long been changed. We now have a separate/new T1 for data and the original T1/channel bank which is in fact only for voice. When I talk with our CLEC (USLEC) I should now expect them to tell me there are only 7 channels provisioned on the Voice T1. (If I understand your comments) Also, Sounds as though I should compare pricing to a PRI ISDN circuit and inquire about our current Voice T1's signaling (I wonder if this is in reference to CLOCKING of a TDM circuit?) We will need a few Analog lines for various IT Dpt. issues therefore a Zapata T1 card for the Asterisk server is out. Also, an earlier reference to 'A port' comes for my reading about the TDM400 cards. That material states the card has 4 'ports' configurable to FXO or FXS type 'ports'. If I understand your comments, 1 Channel of a T1 OR PRI equals 1 port on an TDM400 card or OpenSwitch12 for that matter. Lastly, I have seen recommendations to only install ONE card within an Asterisk server for stability and performance reasons. Are we past that need with todays current baseline server products? Again, Thank you and anyone who replies to this dialog. It is very appreciated in helping raise confidence with this project. -- -Original Message- From: [EMAIL PROTECTED] on behalf of Alex Balashov Sent: Tue 4/10/2007 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices, too, especially if you go shopping with some friendly CLECs. The rule of thumb in the industry is that generally, once you pass the threshold of six or seven POTS lines, it becomes economical to just order an entire PRI, and once you do that, there usually aren't *very* considerable savings to be gained from turning down all but a few channels. A PRI has 23 channels (bearer channels (B channels)) and one signaling channel (D channel); it's a type of T1-based ISDN interface. So, you might potentially be able to get 23 in/outbound phone lines for roughly
Re: [asterisk-users] QSIG configuration
*zapata.conf switchtype=qsig On 4/10/07, George C. Attopany [EMAIL PROTECTED] wrote: Hello, Anyone to help with information on configuring Q.SIG in Asterisk ? I run ASTERISK 1.4 with a Wildcard TE410P-Xilinx on Fedora Core 6 and Zaptel 1.4 I need to tie this ASTERISK system to a Panasonic TDA200 PABX which has ISDN PRI Card which requires QSIG signalling for seamless integration with the ASTERISK system. Any help would be appreciated. Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mounting this task.
James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices, too, especially if you go shopping with some friendly CLECs. The rule of thumb in the industry is that generally, once you pass the threshold of six or seven POTS lines, it becomes economical to just order an entire PRI, and once you do that, there usually aren't *very* considerable savings to be gained from turning down all but a few channels. A PRI has 23 channels (bearer channels (B channels)) and one signaling channel (D channel); it's a type of T1-based ISDN interface. So, you might potentially be able to get 23 in/outbound phone lines for roughly the same cost or a modest increase, which would increase your organisation's capacity to do things like conference calling and other things which tie up large amounts of outside lines. Do beware that if you go this route, PRIs can be ordered as inward-only (typically used for modem and termination-only telephony applications like voicemail, IVR, conferencing, etc.) or bidirectionally. If we go with a Zapata T1 card for the Asterisk server would we be able to provision an analog phone line, for say a FAX machine from it? No, not if the card doesn't have FXS ports on it. But you could get another Digium or Digiumlike card that does, even if it's just a single-port (like the hugely popular X100P, which is very inexpensive) and pull that off. Let me know what else we can answer, or if I substantially misunderstood your question. Good luck, -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
