Re: [asterisk-users] Modem bridging on Asterisk (no VoIP involved)
Hi Alberto, I think that here you can find useful hw: http://www.patapsco.co.uk/ Marino On Jan 23, 2008 9:39 AM, Alberto Pastore <[EMAIL PROTECTED]> wrote: > Hi everybody. > > I know maybe this question has been posted some time ago, but > I need your updated opinion on the subject. > > I'm replacing our old pbx with asterisk. > I have two TE207 dual pri (e1) cards on a clustered system > (one on each node). > > I absolutely need to connect 4/5 analog extensions with > modems, they're being used for remote assistance on very > old systems which cannot be upgraded to native IP links. > > Is there a good hardware that can bridge the e1 lines > on the digium te207 card to my modems? > A PCI card? An external box? > > I don't want to relay modem connections over ip, > I just need to bridge them internally on the asterisk server: > > E1 ==> TE207 ==> Asterisk ==> (some hardware with FXS) ==> modems > > TIA for your replies. > > -- > Alberto Pastore > B-Press Srl - Gruppo MSoft > P.IVA 01697420030 > P.le Lombardia, 4 - 28100 Novara - Italy > Tel. 0321-499508 > Fax 0321-492974 > http://www.msoft.it > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a softswith for a small ISP
Hi Paolo, Unfortunately there are not over the counter solutions. I used OpenSer in order to add scalability (point 1) and a custom web application for point 2. Map On Mon, Feb 11, 2008 at 3:16 PM, Paolo Losi <[EMAIL PROTECTED]> wrote: > Hi all, > > we are considering Asterisk as a possible solution for providing > VoIP services. > The list of requirements would be: > > 1) Load balancing and clustering (horizontal scalability) > 2) User portal for self provisioning > > Googling around doesn't yield any "standard" state-of-art > solutions for points 1 and 2. > > Is anyone using asterisk for this scenario? > Which solution are you using? > > Thanks > Paolo > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference rooms
Hi Fabio, Once you have an Asterisk box that have a conference room configured and a VoIP phone the supports forward you can easily forward your guests to the conference room. Moreover you can create a conference room extension available, via password, from the PSTN line. Hope this can help you. On Nov 13, 2007 3:38 PM, Fabio Cappelletti <[EMAIL PROTECTED]> wrote: > I all, > I have a question about the use of conference rooms: can I, with a Voip > telephone or softphone call some other telephone and invite them in a > conference room? I read a lot of documentations about asterisk, but i > can't find any example ! > > Thanks, best regard > > Fabio > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bridge two connected calls
Hi Alberto, I think that meetme and/or park should work. We done something like this suing queue as well. Could you please let us know why task #6 does not work in your case. Map On Nov 23, 2007 8:38 AM, Alberto Pastore <[EMAIL PROTECTED]> wrote: > Hi everybody. > > I am in the following scenario: > > 1 Customer "A" calls an asterisk box over a Zap channel on > a toll free number during night time > > 2 The incoming call enters an AGI script on the dialplan > > 3 The AGI script plays back a welcome message, then > starts the music-on-hold stream > > 4 The AGI script originates a calls to a > stand-by operator's cell phone (operator "B") > > 5 When the operator "B" answers the call, he is prompted > (via another AGI script in the dialplan) > to dial "1" to be recognized as "human" (the AMD() > function is too random to be useful) > > 6 After being recognized as human, Customer "A" must > be bridged to Operator "B" > > Everything is ok from 1 to 5, but I cannot really figure out > how to accomplish task #6 > I've tried with MeetMe or call parking but with no success. > > Can anyone point me in the right direction? > Thanks > > > -- > Alberto Pastore > B-Press Srl - Gruppo MSoft > P.IVA 01697420030 > P.le Lombardia, 4 - 28100 Novara - Italy > Tel. 0321-499508 > Fax 0321-492974 > http://www.msoft.it > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Remove a TDM Card
Hi, This procedure should work whether you have remove some "VoIP" card from your VoIP box. Anyway be careful On Nov 29, 2007 11:14 AM, Sasa <[EMAIL PROTECTED]> wrote: > Hi, my problem isn't on new voip box with latest asterisk version...my > problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this > steps for remove rightly TDM Card: > > - remove line configuration about tdm card in zapata.conf and zaptel.conf > - remove in rc.modules and rc.modules-2.4.33.3 line: > /sbin/modprobe wctdm24xxp && /sbin/ztcfg -vv > - rmmod wctdm24xxp > - halt > - remove physically card tdm from pc (box voip 1) > - restart box voip 1 > > ..this procedure is ok ? > Thanks ! > > -- > > Salvatore. > > > > - Original Message - > From: "Tzafrir Cohen" <[EMAIL PROTECTED]> > To: > Sent: Thursday, November 29, 2007 1:50 AM > Subject: Re: [asterisk-users] Fw: Remove a TDM Card > > > > On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote: > >> Hi, sorry but perhaps I don't have explained clearly my problem...now I > >> have > >> a box voip that must be replace with another box voip but I want to do > >> test > >> before remove the old voip from production. > > > > With later versions of Zaptel you have zapconf and genzaptelconf . Use > > either of them to generate /etc/zaptel.conf and to generate a sample > > zapata.conf snippet in /etc/asterisk/zapata-channels.conf . > > > > -- > > Tzafrir Cohen > > icq#16849755 jabber:[EMAIL PROTECTED] > > +972-50-7952406 mailto:[EMAIL PROTECTED] > > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outgoing calls with Digium B410P
Hi Daniele, Could you please tell us what exactly happens? Are your able to see some error in the log/console? On Jan 7, 2008 11:53 AM, daniele visaggio <[EMAIL PROTECTED]> wrote: > Hi all! > > Sorry for my poor english, i'm italian. > > I installed Digium B410P on my asterisk server. I followed the official > installation instructions found on digium site. These instruction, in my > opinion, are not clear, so i tried with other ways (found on the trixbox > site). I found this: > http://www.trixbox.org/forums/trixbox-forums/trunks/howto-install-script-digium-b410p > > Now i can receive calls from pstn, but i can't do any call to pstn. > > I attach my /etc/misdn-init.conf/ and also my etc/asterisk/misdn.conf > file for clarity. > > Thanks - Daniele > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outgoing calls with Digium B410P
Daniele,
you need an "external calls" rule in your extension.conf, that is 1 to call
using PSTN line.
Please send your extension and we can take a look to find your problem.
p.s.
I'm Italian too.
On Jan 7, 2008 3:03 PM, daniele visaggio <[EMAIL PROTECTED]> wrote:
> Hi Daniele,
> >
> > Could you please tell us what exactly happens?
> > Are your able to see some error in the log/console?
> >
> >
> > Thanks for your answer.
>
> I'm managing the asterisk server from a windows client via ssh (putty
> client), so i can't paste here the output of the asterisk CLI, but when i
> try to do a call to the PSTN i see a lot o f messages, but no one of them
> looks like an error. They are of this type:
>
> -- Executing [EMAIL PROTECTED]:1] Macro ("SIP/501-0827c9e82,
> dialout-trunk|1|0266200xxx||") in new stack
>
> 0266200xxx is the number i'm trying calling to, 501 is my extension number
> (i have one SIP hard-phone); the number 1 before 0266200xxx is part of the
> dial patterns i created, because i want to dial 1 before the outgoing
> number.
>
> Anyway, on the official digium documentation, it's written that "in order
> to call out over a port, the Dial () command has to be formatted as follows:
> Dial (misdn/g:myoutsidelines/ $ {EXTEN}). Have i to edit my extension.confand
> insert in a dial command to do an outgoing call?
>
> Thanks a lot - Daniele
>
> ___
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] no outgoing calls with Digium B410P
Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration -> Window -> Lines of scrollback and put a number greater than 200 :-). I suggest 10. On Jan 7, 2008 4:00 PM, daniele visaggio <[EMAIL PROTECTED]> wrote: > 2008/1/7, map <[EMAIL PROTECTED]>: > > > > Daniele, > > you need an "external calls" rule in your extension.conf, that is 1 to > > call using PSTN line. > > > > Please send your extension and we can take a look to find your problem. > > > > p.s. > > I'm Italian too. > > > > Ok, i attach my extension.conf. > > Thank you very much, i'm very happy of finding another italian asterisk > user. > > Ciao e grazie! > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video phones on 1.4.7
Hi think that once SIP/SDP invite/reinvite is sent you can not change to video stream. On 7/10/07, Jerry Geis <[EMAIL PROTECTED]> wrote: I have 3 phones P1 is a non video phone - grandstream P2 is a Grandstream GXV3000 P3 is a Grandstream GXV3000 Using P1 to place a call to P2 I get audio only (as expected). Then on P1 I transfer the call to P3 and I still only get audio. At this point shouldn't the two video phones P2 and P3 say to each other we are video and so startup the video stream??? This is not working at this time? OR is there something I am missing. Thanks, Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help: H323 and SIP
Hi Alex, You should create a dial plan to route sip calls to H.323 calls. Take a look at : http://www.voip-info.org/wiki/ On 8/6/07, Alessandro Russo <[EMAIL PROTECTED]> wrote: > > Hi to all, > I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. > I've tested h323 using ohphone and I can talk between them, then I've > tested SIP with Twinkle softphones and function very well. > Now I have to perform call from h323 to sip and viceversa. > How can I do it > I receive h323 call from a Cisco Voice GW to my Asterisk and this call > have to go to a SIP phone: > - PSTN ==> CiscoVoiceGW(h323) ==> Asterisk ==> SIP > - SIP ==> Asterisk ==> CiscoVoiceGW(h323) ==> PSNT > > I've now idea how to configure asterisk (conf file) and softphones... > Thanks for all! > > -- > AxR. > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help: H323 and SIP
Hi Alex, you should have a "route" for each extensions you would like to reach in your extension.conf file. Dial Plan is the main concept to understand in Asterisk. Feel free to send you conf and I'll take a look. On 8/6/07, Alessandro Russo <[EMAIL PROTECTED]> wrote: > > Hi, > thanks for reply > I'm reading more about Dialplan, but until now, I've not found > anything...(like example or tutorial) > With the word "route" you are intending the "Goto" command?? > Please spent some minutes for helping me ^_^ > If you are agree, I send you some information about configuration files. > Thx > > > On 8/6/07, map < [EMAIL PROTECTED]> wrote: > > > > Hi Alex, > > > > You should create a dial plan to route sip calls to H.323 calls. > > > > Take a look at : > > http://www.voip-info.org/wiki/ > > > > > > > > > > On 8/6/07, Alessandro Russo <[EMAIL PROTECTED]> wrote: > > > > > Hi to all, > > > I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. > > > I've tested h323 using ohphone and I can talk between them, then I've > > > tested SIP with Twinkle softphones and function very well. > > > Now I have to perform call from h323 to sip and viceversa. > > > How can I do it > > > I receive h323 call from a Cisco Voice GW to my Asterisk and this call > > > have to go to a SIP phone: > > > - PSTN ==> CiscoVoiceGW(h323) ==> Asterisk ==> SIP > > > - SIP ==> Asterisk ==> CiscoVoiceGW(h323) ==> PSNT > > > > > > I've now idea how to configure asterisk (conf file) and softphones... > > > Thanks for all! > > > > > > -- > > > AxR. > > > ___ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > Alessandro R. > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unable to register to tnet.it
Hi Giorgio, Do you have any log showing some error? Did you already have a look at SIP connection messages from and to this SIP server? I suggest you to use wireshark to check sip messages. Thanks, Marino On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo < [EMAIL PROTECTED]> wrote: > Hi, > I cannot make my Asterisk register to tnet.it, an italian SIP provider. > I tried many register string formats and tried to set realm and > outboundproxy (sip.tnet.it) too but without any result. > Still I cannot register (but for example messagenet works fine). > Is there anybody who tried this provider and successfully registered to it? > > Thank you. > > Giorgio. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unable to register to tnet.it
Hi Giorgio, >From your email seems clear that your Asterisk box can not resolve tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < [EMAIL PROTECTED]> wrote: > Hi Marino, > Asterisk gives a timeout on registration and a "no such host" because > cannot resolve tnet.it but that server address is not resolvable so I > think that is not a problem (my zoiper connects to the provider without > problems, so why shouldn't Asterisk??) > Activating "sip debug" shows the register packets but nothing in return. > I used the proxy tnet gave me but nothing changes. > Searched on their site for some help about Asterisk configuration but > nothing...the same on the rest of internet. > > Giorgio > > > map wrote: > > Hi Giorgio, > > > > Do you have any log showing some error? > > Did you already have a look at SIP connection messages from and to > > this SIP server? I suggest you to use wireshark to check sip messages. > > > > Thanks, > > Marino > > > > On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo > > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > > wrote: > > > > Hi, > > I cannot make my Asterisk register to tnet.it <http://tnet.it>, an > > italian SIP provider. > > I tried many register string formats and tried to set realm and > > outboundproxy (sip.tnet.it <http://sip.tnet.it>) too but without > > any result. > > Still I cannot register (but for example messagenet works fine). > > Is there anybody who tried this provider and successfully > > registered to it? > > > > Thank you. > > > > Giorgio. > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unable to register to tnet.it
Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. >From your previous email : ... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo < [EMAIL PROTECTED]> wrote: > Hi Marino, > > I tried to connect zoiper directly to the provider with the same account > parameters I'm using with Asterisk. Zoiper connects without problems. It > is true tnet.it is not resolvable but I can use the proxy URL > sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm > trying to understand where is the problem. I thought I had to specify > the outboundproxy parameter in the general section of sip.conf to make > Asterisk correctly work but it seems that's not enough. > > > Thank you. > > Giorgio > > > map wrote: > > Hi Giorgio, > > > > From your email seems clear that your Asterisk box can not resolve > > tnet.it <http://tnet.it> and SIP register messages are not replied. > > I suggested to check if your Asterisk box is really sending SIP > > messages, you can use a net sniffer. > > Did you alerady used different sip client with the same sip account of > > your Asterisk box? > > > > Did you use zoiper from the same box? > > > > Marino > > > > p.s. > > Are you Italian? > > > > > > On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo > > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > > wrote: > > > > Hi Marino, > > Asterisk gives a timeout on registration and a "no such host" because > > cannot resolve tnet.it <http://tnet.it> but that server address is > > not resolvable so I > > think that is not a problem (my zoiper connects to the provider > > without > > problems, so why shouldn't Asterisk??) > > Activating "sip debug" shows the register packets but nothing in > > return. > > I used the proxy tnet gave me but nothing changes. > > Searched on their site for some help about Asterisk configuration but > > nothing...the same on the rest of internet. > > > > Giorgio > > > > > > map wrote: > > > Hi Giorgio, > > > > > > Do you have any log showing some error? > > > Did you already have a look at SIP connection messages from and to > > > this SIP server? I suggest you to use wireshark to check sip > > messages. > > > > > > Thanks, > > > Marino > > > > > > On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo > > > <[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]> > > <mailto:[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>>> > > > wrote: > > > > > > Hi, > > > I cannot make my Asterisk register to tnet.it > > <http://tnet.it> <http://tnet.it>, an > > > italian SIP provider. > > > I tried many register string formats and tried to set realm and > > > outboundproxy (sip.tnet.it <http://sip.tnet.it> > > <http://sip.tnet.it>) too but without > > > any result. > > > Still I cannot register (but for example messagenet works > fine). > > > Is there anybody who tried this provider and successfully > > > registered to it? > > > > > > Thank you. > > > > > > Giorgio. > > > > > > ___ > > > -- Bandwidth and Colocation Provided by > > http://www.api-digital.com -- > > > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > >
Re: [asterisk-users] Asterisk unable to register to tnet.it
Hi Giorgio, RE my point 2: You should test a sip client, whatever you want, on your linux/asterisk box just to double check that this box works fine. If you are abel to connect with a sip client from tour asterisk box we will be sure that the network configuration is ok. You have no natt but maybe your routing table is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo < [EMAIL PROTECTED]> wrote: > Hi Marino, > > 1) yes I can connect using the account > 2) no, I'm running zoiper on a different machine. I'm using an Asterisk > server which is not behind nat as for the machine zoiper is runnin' on. > The Asterisk server is directly connected to internet, I wanted to avoid > nat problems, that's why. > Moreover I tried to create a simpler account on my zoiper using > username, password and domain name only and it works even without > setting the sip proxy. > I changed the Asterisk server too: now I'm using a test one where I can > ping tnet.it from... but nothing changes. > I'm using this string: > register => 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 > I changed it in many other forms following the wiki pages but nothing. > I see sip packets are sent to tnet.it (I set up sip debug) but I always > get this message: > > Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- > Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) > > I wonder why I had no problems with the other provider we are using > while tnet.it is making me get crazy > > Thank you. > > Giorgio > > > map wrote: > > Hi Giorgio, > > > > Just to recap: > > 1) you are able to connect to tnet.it <http://tnet.it> by using the > > same account of your asterisk box. There is no issue related to your > > account. > > 2) Could you please confirm that you are running zoiper from the same > > box used by asterisk? If yes we can exclude some generic network issues. > > > > > > From your previous email : > > ... > > Activating "sip debug" shows the register packets but nothing in return. > > ... > > > > I think that this is a network related issue, but you have to solve it > > by using a Asterisk config file. > > > > Unfortunately I think that the faster way to solve your problem is > > trying to understand if sip messages are correctly sent to tnet. > > I strongly suggest to use http://www.wireshark.org/ previoulsly named > > Ethereal in order to check sip messages. > > I have to sniff both asterisk and zoiper sip messages. > > I know that this can be tricky but this can help you to understand > > what is wrong in sip messages. > > > > Please let me know if you need more detail. > > > > > > Marino > > > > On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo > > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > > wrote: > > > > Hi Marino, > > > > I tried to connect zoiper directly to the provider with the same > > account > > parameters I'm using with Asterisk. Zoiper connects without > > problems. It > > is true tnet.it <http://tnet.it> is not resolvable but I can use > > the proxy URL > > sip.tnet.it <http://sip.tnet.it> which seems to work with Zoiper > > but not with Asterisk. I'm > > trying to understand where is the problem. I thought I had to specify > > the outboundproxy parameter in the general section of sip.conf to > make > > Asterisk correctly work but it seems that's not enough. > > > > > > Thank you. > > > > Giorgio > > > > > > map wrote: > > > Hi Giorgio, > > > > > > From your email seems clear that your Asterisk box can not resolve > > > tnet.it <http://tnet.it> <http://tnet.it> and SIP register > > messages are not replied. > > > I suggested to check if your Asterisk box is really sending SIP > > > messages, you can use a net sniffer. > > > Did you alerady used different sip client with the same sip > > account of > > > your Asterisk box? > > > > > > Did you use zoiper from the same box? > > > > > > Marino > > > > > > p.s. > > > Are you Italian? > > > > > > > > > On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo > > > <[EMAIL PROTECTED] >
Re: [asterisk-users] IP door opening devices
Hi all, maybe there is no opener device at all. Anyway take a look here : http://www.barix.com/ On Thu, Jul 24, 2008 at 12:11 PM, Gordon Henderson < [EMAIL PROTECTED] <[EMAIL PROTECTED]>> wrote: > On Thu, 24 Jul 2008, Chris Bagnall wrote: > > > Greetings list, > > > > We have a client with an analogue door intercom/opening unit which we're > > attempting to replace with an IP variant. The existing unit has the > > following functionality: > > > > 1) Intercom - visitor hits "call", talks to operator > > > 2) Door opening - operator can open the door by dialling a 4-digit PIN > > followed by * (the door unit interprets the DTMF tones) > > > 3) Door opening - the door unit has a numeric keypad to enable approved > > persons to enter by entering the 4-digit PIN on the keypad > > > > We've tried getting the existing unit working with an ATA, but it's only > > about 50% reliable (hangup not always detected, DTMF not always > > detected, etc.), so it's probably time to look at fully IP alternatives. > > > > Any suggestions gratefully appreciated. > > There was talk of this a week or 2 ago on the list - look into the > archives. I don't think there was anything that successfull though... > > I have to say though - if you have such an integrated unit that needs > nothing more than an analogue connection (and power, presumably), I'd love > to know the make - for me, (or rather one of my clients) it would be > worthwhile trying to find an ATA that would work with it.. > > Got a name/website for the opener device? > > Gordon > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Hi suggest this : http://www.patapsco.co.uk Maybe a little bit more expensive but a very good product. Map On Thu, Aug 28, 2008 at 3:08 AM, George Pajari <[EMAIL PROTECTED]>wrote: > Why a three-port PRI card? > > Just put a two-port card into your Asterisk server, pull off those DIDs > you want to process locally, and send the rest over the second port to > the PBX. In the reverse direction, intercept calls from the PBX to the > Asterisk DIDs but pass everything else to the telco. > > We just finished installing just a system for a car dealership in BC > that is splitting the body shop off into a separate building running off > Asterisk while the rest of the company remains on their existing legacy > PBX for a while longer (they'll come over later). > > g. > > Jeremy Mann wrote: > > > Does anyone know of a pri splitter device? Something that would take > > an incoming PRI, and based on DID send that out one of other multiple > > PRI ports? > > > > I'm needing to take a single PRI from the telco, and send it to two > > separate phone systems(one asterisk) based on DID. > > > > I know I could probably achieve the same thing with a 3 port PRI card > > in a server, but I'd like something braindead easy to configure from > > both a hardware and software perspective. > > > > -- > George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 > www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca >www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca > Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 protocol
Yes you can. Obviously you have to compile, configure and add chan_h323 to Asterisk. Map On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman <[EMAIL PROTECTED]>wrote: > hi. > > i have two IP phones that are in H323 protocol. How can i test that > these two phones are working? For IP phone (SIP) i used asterisk > server. can i use asterisk server to test the ip phone with H323 > protocol. > > -- > Mahboob Zaman > System Engr > Systems & Services Limited > Cell: +8801712280308 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't hear any sound (This time in plain text)
If you're in the same lan, I think taht you have some problem with the codec. Doulbe check codecs config both in xlite and in sip.conf. On 3/9/07, Asterisk Asterisk <[EMAIL PROTECTED]> wrote: Hey, I am a new to asterisk and softphones. Ihave recently installed and configured linux and 2 xlite clients all in linux fedora core 6. I have also made a dial plan for the two users. But when i dial from one xlite client to another i can hear the ring tone but when i answer the call i can not hear any sound. I have checked my microphone and its working fine. Please could anyone help me on this issue. Sorry,I did not know that the list could not open any attachments. Regards Szabistians Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] extra field
Hi, Could you please explain what your provider is expecting? You should only have to provide your public IP address. On 4/4/07, Il Neofita <[EMAIL PROTECTED]> wrote: Hi, I am using my asterisk server like a gateway and one provider ask me to pass an extra field with the IP of the peer that is using the connection, probably to have more control on the authentication. I was wondering how I can implement this. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Linksys SPAs work well with Asterisk On 4/13/07, Luca Corti <[EMAIL PROTECTED]> wrote: On Thu, 2007-04-12 at 14:47 -0400, J. Oquendo wrote: > 1) Snom > 2) none! (they're all pretty much the same to me) > 3) none! (they all have their pros and cons) > 4) Cisco > 5) ASStra > 6) Polycrud You haven't even mentioned Linksys SPAs. Have you tested them? ciao Luca ___ --Bandwidth 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] Video phone settings???
Hi,
It seems that you are using different audio codec (Unknown RTP codec
96 received)
Try to use standard audio code. Sometimes telephone use codec with bad
rtp code inside. I use alw or ulaw for my test.
Marino
On 7/11/05, Ronald_Wiplinger <[EMAIL PROTECTED]> wrote:
> Giorgio Incantalupo wrote:
>
> > Hi,
> > try videosupport=yes in the general section of sip.conf. Maybe it can
> > work.
>
>
> I have already set that. Without that NO video at all at any try.
>
>
> bye
>
> Ronald
>
> >
> > Giorgio.
> >
> >
> > Ronald_Wiplinger wrote:
> >
> >> I have three video phones here for testing:
> >>
> >> Extension 6003 is Eyebeam
> >> Extension 6004 is a hard phone (model 8770)
> >> Extension 6005 is a hard phone (model 8882)
> >>
> >> Can anybody have a look at my settings and the output I get from all
> >> kinds of dialings, please.
> >>
> >> The sip settings for all phones is (user / password different):
> >>
> >> [6003]
> >> type=friend
> >> username=6003
> >> secret=pwd
> >> qualify=200
> >> nat=yes
> >> host=dynamic
> >> canreinvite=yes
> >> context=from-sip
> >> callerid=Ronald Wiplinger <6003>
> >> dtmfmode=rfc2833
> >> disallow=all
> >> allow=ulaw
> >> allow=alaw
> >> allow=h261
> >> allow=h263
> >> allow=h263p
> >>
> >>
> >>
> >>
> >>
> >>
> >> Tests on 7/11/2005
> >>
> >> Eybeam to 8770
> >>
> >> both screens are black!!!
> >>
> >>
> >> e*CLI>
> >>-- Executing Dial("SIP/6003-94ec", "SIP/6004|60|trm") in new stack
> >>-- Called 6004
> >>-- Started music on hold, class 'default', on SIP/6003-94ec
> >>-- SIP/6004-4b4d is ringing
> >>-- SIP/6004-4b4d answered SIP/6003-94ec
> >>-- Stopped music on hold on SIP/6003-94ec
> >>-- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d
> >> == Spawn extension (from-sip, 6004, 1) exited non-zero on
> >> 'SIP/6003-94ec'
> >>
> >>
> >>
> >> --
> >>
> >> Eybeam to 8882
> >>
> >> both screens are black!!!
> >>
> >>
> >> *CLI>
> >>-- Executing Dial("SIP/6003-8a2e", "SIP/6005|60|trm") in new stack
> >>-- Called 6005
> >>-- Started music on hold, class 'default', on SIP/6003-8a2e
> >>-- SIP/6005-fa6a is ringing
> >>-- SIP/6005-fa6a answered SIP/6003-8a2e
> >>-- Stopped music on hold on SIP/6003-8a2e
> >>-- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a
> >> == Spawn extension (from-sip, 6005, 1) exited non-zero on
> >> 'SIP/6003-8a2e'
> >>
> >>
> >>
> >> --
> >>
> >> 8770 to 8882
> >>
> >> both screens are black!!!
> >>
> >>
> >> *CLI>
> >>-- Executing Dial("SIP/6004-5e88", "SIP/6005|60|trm") in new stack
> >>-- Called 6005
> >>-- Started music on hold, class 'default', on SIP/6004-5e88
> >>-- SIP/6005-5180 is ringing
> >>-- SIP/6005-5180 answered SIP/6004-5e88
> >>-- Stopped music on hold on SIP/6004-5e88
> >>-- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> == Spawn extension (from-sip, 6005, 1) exited non-zero on
> >> 'SIP/6004-5e88'
> >>
> >>
> >>
> >> --
> >>
> >> 8770 to Eyebeam
> >>
> >> 8770 gets picture, Eybeam no picture
> >>
> >>
> >>-- Executing Dial("SIP/6004-5e88", "SIP/6005|60|trm") in new stack
> >>-- Called 6005
> >>-- Started music on hold, class 'default', on SIP/6004-5e88
> >>-- SIP/6005-5180 is ringing
> >>-- SIP/6005-5180 answered SIP/6004-5e88
> >>-- Stopped music on hold on SIP/6004-5e88
> >>-- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP
> >> codec 96 received
> >> == Spawn extension (from-sip, 6005, 1) exited non-zero on
> >> 'SIP/6004-5e88'
> >>-- Executing Dial("SIP/6004-2cff", "SIP/6003|60|trm") in new stack
> >>-- Called 6003
> >>-- Started music on hold, class 'default', on SIP/6004-2cff
> >>-- SIP/6003-322c is ringing
> >>-- SIP/6003-322c answered SIP/6004-2cff
> >>-- Stopped music on hold on SIP/6004-2cff
> >>-- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c
> >> == Spawn extension (from-sip, 6003, 1) exited non-zero on
> >> 'SIP/6004-2cff'
> >>
> >> --
> >>
> >> 8882 to Eyebeam
> >>
> >> both screens are black!!!
> >>
> >>
> >>-- Executing Dial("SIP/6005-3361", "SIP/6003|60|trm") in new stack
> >>-- Called 6003
> >>-- Started music on hold, class 'default', on SIP/6005-3361
> >>-- SIP/6003-9ed0 is ringing
> >>-- SIP/6003-9ed0 answered SIP/6005-3361
> >>-- Stopped music on hold on SIP/6005-3361
> >>-- Attempting nati
