[asterisk-users] Sangoma U100
I am having problems trying to get a Sangoma U100 USB FXO adapter working with Asterisk. After all the configuration Asterisk (1.6.2.0-rc2) detects the port and I can even call in without a problem (after making sure no echo canceller is configured for that channel). The problem is that when I try to call out it never works. The channel hangs up immediately. I cannot even use the originate CLI command to call out. Anyone has experience with these adapters? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ -- 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] Setting up skype
Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well ? If so, why do I have the problem ? And would this affect local channels as well ? Julian 2009/12/6 Kevin P. Fleming : > Julian Lyndon-Smith wrote: > >> external => ddi => dial(skype) >> >> and got a load of static with >> >> WARNING[15328]: channel.c:3098 set_format: Unable to find a codec >> translation path from 0x100 (g729) to 0x8 (alaw) >> >> on the console. >> >> Fired up a sip client, made the same call, and all was ok. >> >> Any clues ? > > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for > nearly all calls, so handling calls via those paths requires a G.729 > transcoder on the system if the target of the call will not also be > using G.729. This is why the Skype For Asterisk license includes > licenses for Digium's G.729 software transcoder as well. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- 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 to Email
On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
> And, then send an email to the party. Example
>
> 3035551...@tmobile.net
>
> Summary
> 1. Capture the CID number.
> 2. Prepend his number to his service provider SMTP address
> 3. Email it to his phone
System(echo body of message | mail -s "subject line"
${the_caller_...@tmobile.net)
Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
1.4.x and below.
>
> I assume I need to install SendMail and play around with CID stuff.
Sendmail, postfix, exim, qmail - any program that provides a local
sendmail interface.
I personally prefer postfix.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/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
[asterisk-users] Can Asterik act as a SIP Proxy
Hi, I am new to Asterisk. Can you pl clarify me the following basic questions I have: 1. Can I use Asterisk as a SIP Proxy. ( I want it to act as proxy not a B2b/GW) 2. If so, will it act as stateless (or) stateful 3. I did see the website, I could not find the info: is there a documentation which can be a starting point f me to understand how it can be made to use as Proxy serve.and how to use Astrisk. If this is not the forum , pl advice me to the correct forum. TIA, Gayatri ___ -- 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] G729: TC400B vs Software Encoder
We run asterisk on c2d 2.66GHz so looks like we should be fine with software encoder. Just to give you more details, below is the relevant row from the translation table (using digium g.729 software encoder). Any estimate on how many simultaneous calls we should be able to make using g.729 and ilbc code without taxing quality. > core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g.729 iLBC slin - 2999 - 7999 Thanks Jim On Sat, Dec 5, 2009 at 8:59 PM, Tilghman Lesher wrote: > On Saturday 05 December 2009 08:25:53 Jim Boykin wrote: >> Thanks. What do you suggest for system with 15 max simultaneous calls? > > As long as your machine is at least a single CPU, 2GHz or better, you should > be fine with the software encoder. The TC400 is really only necessary on > systems where the transcoding load alone would overtax the CPU. It's really > difficult to estimate a good number of where you should start considering the > TC400, because there are so many other tasks where the CPU may be required > and would affect the overall result, but the number is probably somewhere > north of 40 simultaneous calls. > > I'm sure somebody will chime in and say they're using 100 simultaneous calls > on the software encoder alone. As I said, the number is extremely fungible, > based upon what else the CPU is doing. > > -- > Tilghman Lesher > Digium, Inc. | Senior Software Developer > twitter: Corydon76 | IRC: Corydon76-dig (Freenode) > Check us out at: www.digium.com & www.asterisk.org > > ___ > -- 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] Setting up skype
On Saturday 05 December 2009 20:34:58 Sam wrote: > Somewhat off topic but does anyone know if the price for the license > will go down in the future? It seems strange that I can use skype for > free on my computer but to put it on asterisk cost $66... It's cheaper if you go through a Digium reseller. List is $60, and resellers may sell it for less than that. It's 10% more expensive on the website, because we want to encourage you to go through resellers. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org ___ -- 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] Setting up skype
Somewhat off topic but does anyone know if the price for the license will go down in the future? It seems strange that I can use skype for free on my computer but to put it on asterisk cost $66... Sam ___ -- 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
[asterisk-users] Asterisk to Email
How can this scenario be implemented please? THIS IS NOT A SEND TEXT application. A call arrives on the IVR. After hearing several vectors to guide the person through information I want to confirm a transaction via email to his cell phone. Specifically, I want to use his phone number and then append the SMTP suffix from his service provider. Press 1 if you use Verizon, 2 if you use ATT, 3 if you use Sprint, 4 for T-Mobile, etc. And, then send an email to the party. Example 3035551...@tmobile.net Summary 1. Capture the CID number. 2. Prepend his number to his service provider SMTP address 3. Email it to his phone I assume I need to install SendMail and play around with CID stuff. Any hints? ___ -- 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] Setting up skype
Julian Lyndon-Smith wrote: > external => ddi => dial(skype) > > and got a load of static with > > WARNING[15328]: channel.c:3098 set_format: Unable to find a codec > translation path from 0x100 (g729) to 0x8 (alaw) > > on the console. > > Fired up a sip client, made the same call, and all was ok. > > Any clues ? The clues are in the documentation; SkypeIn and SkypeOut use G.729 for nearly all calls, so handling calls via those paths requires a G.729 transcoder on the system if the target of the call will not also be using G.729. This is why the Skype For Asterisk license includes licenses for Digium's G.729 software transcoder as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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 use SIP hints and BLF for realtime extensions on Aastra phones?
It is already enabled in sip.conf. On Sat, Dec 5, 2009 at 12:16 PM, Carlos Chavez wrote: > You need to enable rtcachefriends=yes in sip.conf > > *On Fri, 4 Dec 2009 23:57:24 -0500, Zeeshan Zakaria wrote* > > Hi, > > > > I need to make use of BLF feature on Aastra 6757i phones but its an > Asterisk 1.4 using realtime architecture. Extensions are defined in realtime > database and dial plan is in AEL. I am able to correctly setup hints in the > dialplan, but they don't work. Did some research and found out that hints > don't work work with realtime extensions. Is there any work around? > > > > On voip-info I read that Snom phones can use BLF without using hints. Is > it possible to do similar on Aastra phones? > > > > Any guidance will be highly appreciated. > > > > -- > > Zeeshan A Zakaria > > > -- > Carlos Chavez > Director de Tecnología > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > ___ > -- 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 > -- Zeeshan A Zakaria ___ -- 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] Setting up skype
well, aint that a bugger. Just looked at my contacts list on skype and
the bloody thing is working ... wtf ?
Another question: I just tried calling in like this
external => ddi => dial(skype)
and got a load of static with
WARNING[15328]: channel.c:3098 set_format: Unable to find a codec
translation path from 0x100 (g729) to 0x8 (alaw)
on the console.
Fired up a sip client, made the same call, and all was ok.
Any clues ?
2009/12/5 Julian Lyndon-Smith :
> As I have no friends and no life I thought that I would set up my
> asterisk server with Skype.
>
> 1) Paid the $, got the licence, built and installed
> 2) create a business skype account (called company "foo")
> 3) created a member of the business called "bar"
> 4) updated the skype conf file
> 5) restarted asterisk
>
>
>
> => skype show settings
> Skype For Asterisk Settings:
> engine_directory: /tmp
> data_directory: /var/spool/asterisk/skype
> defaultuser: bar
> bind_address: 0.0.0.0
> bind_port: 0
> rtp_address: 127.0.0.1
> https_proxy:
> https_proxy_user:
> https_proxy_password:
> socks5_proxy:
> socks5_proxy_user:
> socks5_proxy_password:
> disable_tcpauto: no
> disable_udp: no
> debug: no
>
> => skype show users
> Skype Users>
> bar: Logged In
>
> 6) added a test to extensions.conf
>
> exten => 123650,1,Dial(Skype/b...@my.personal.skype)
> exten => 123650,2,Hangup()
>
> and get a
>
> Everyone is busy/congested at this time (1:0/0/1)
> [Dec 5 20:15:08] -- Executing [123...@isdnspan1:2]
> Hangup("Zap/1-1", "") in new stack
>
> My skype client can find "bar", but it is "offline", so I can't place
> calls either
>
> Anyone know what I am doing wrong ?? (1.4 source svn trunk)
>
> TIA
>
> Julian
>
___
-- 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
[asterisk-users] Setting up skype
As I have no friends and no life I thought that I would set up my
asterisk server with Skype.
1) Paid the $, got the licence, built and installed
2) create a business skype account (called company "foo")
3) created a member of the business called "bar"
4) updated the skype conf file
5) restarted asterisk
=> skype show settings
Skype For Asterisk Settings:
engine_directory: /tmp
data_directory: /var/spool/asterisk/skype
defaultuser: bar
bind_address: 0.0.0.0
bind_port: 0
rtp_address: 127.0.0.1
https_proxy:
https_proxy_user:
https_proxy_password:
socks5_proxy:
socks5_proxy_user:
socks5_proxy_password:
disable_tcpauto: no
disable_udp: no
debug: no
=> skype show users
Skype Users>
bar: Logged In
6) added a test to extensions.conf
exten => 123650,1,Dial(Skype/b...@my.personal.skype)
exten => 123650,2,Hangup()
and get a
Everyone is busy/congested at this time (1:0/0/1)
[Dec 5 20:15:08] -- Executing [123...@isdnspan1:2]
Hangup("Zap/1-1", "") in new stack
My skype client can find "bar", but it is "offline", so I can't place
calls either
Anyone know what I am doing wrong ?? (1.4 source svn trunk)
TIA
Julian
___
-- 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
[asterisk-users] Echo issue
I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. Thank you! ___ -- 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 use SIP hints and BLF for realtime extensions on Aastra phones?
You need to enable rtcachefriends=yes in sip.conf On Fri, 4 Dec 2009 23:57:24 -0500, Zeeshan Zakaria wrote > Hi, > > I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk > 1.4 using realtime architecture. Extensions are defined in realtime database > and dial plan is in AEL. I am able to correctly setup hints in the dialplan, > but they don't work. Did some research and found out that hints don't work > work with realtime extensions. Is there any work around? > > On voip-info I read that Snom phones can use BLF without using hints. Is it > possible to do similar on Aastra phones? > > Any guidance will be highly appreciated. > > -- > Zeeshan A Zakaria -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ -- 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] MWI count wrong when using IMAP and VM
any ideas ? or file a bug ? Best Regards, - "--[ UxBoD ]--" wrote: | as soon as I delete the two messages I receive in the console :- | | [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP | Warning: Unknown message data: 1 EXPUNGE | [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP | Warning: Unknown message data: 1 EXPUNGE | | Best Regards, | | | - "--[ UxBoD ]--" wrote: | | | [r...@voip ~]# asterisk -V | | Asterisk 1.6.1.11 | | | | When using the above version with IMAP VoiceMail integration when I | | leave a message my SNOM360 it shows 2 message waiting; yet when | | running voicemail show users from the Asterisk CLI it correctly | | reports 1. | | | | It would appear that when the VM is temporarily stored, and the VM | is | | delivered by IMAP to the remote mail account, the MWI is being | | initiated with a incorrect count. | | | | I then delete the VM from either 1) the phone 2) the mail account | the | | MWI goes blank and the message count shows 0 correctly. | | | | I am still trying to debug but any thoughts on this ? | | | | Here is how I have voicemail.conf :- | | | | [general] | | format=wav49 | | maxsecs=180 | | minsecs=5 | | skipms=3000 | | maxsilence=3 | | silencethreshold=128 | | maxlogins=3 | | imapserver= | | imapfolder=VoiceMail Office | | imapport=993 | | imapflags=ssl | | authuser= | | authpassword= | | | | [voicemail] | | 1001 => 1234imapuser= | | | | Best Regards, | | | | | | -- | | This message has been scanned for viruses and | | dangerous content and is believed to be clean. | | | | SplatNIX IT Services :: Innovation through collaboration | | | | | | ___ | | -- 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 | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- 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 -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] G729: TC400B vs Software Encoder
On Saturday 05 December 2009 08:25:53 Jim Boykin wrote: > Thanks. What do you suggest for system with 15 max simultaneous calls? As long as your machine is at least a single CPU, 2GHz or better, you should be fine with the software encoder. The TC400 is really only necessary on systems where the transcoding load alone would overtax the CPU. It's really difficult to estimate a good number of where you should start considering the TC400, because there are so many other tasks where the CPU may be required and would affect the overall result, but the number is probably somewhere north of 40 simultaneous calls. I'm sure somebody will chime in and say they're using 100 simultaneous calls on the software encoder alone. As I said, the number is extremely fungible, based upon what else the CPU is doing. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org ___ -- 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] G729: TC400B vs Software Encoder
On 12/05/2009 10:19 PM, Kevin P. Fleming wrote: > Jim Boykin wrote: > >> Can someone help me decide between TC400B vs Software Encoder. TC400B >> is really expensive and would like to get opinion if it is worth >> considering. Maximum simultaneous call we handle is hardly 10-15. Will >> going to TC400B improves quality. Any other option then TC400B. >> > All G.729 encoders produce the same quality, since they all do the same > thing. The only difference between various encoders is the amount of > load they play on the system CPU, cost, licensing models, etc. > That's not strictly true. All fixed point G.729 encoders should produce a bit exact output that will match the ITU test vectors. However, most G.729 encoders on PCs use floating point arithmetic, and there is some variability in the output. It should be a pretty small difference, though. Steve ___ -- 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] G729: TC400B vs Software Encoder
Jim Boykin wrote: > Can someone help me decide between TC400B vs Software Encoder. TC400B > is really expensive and would like to get opinion if it is worth > considering. Maximum simultaneous call we handle is hardly 10-15. Will > going to TC400B improves quality. Any other option then TC400B. All G.729 encoders produce the same quality, since they all do the same thing. The only difference between various encoders is the amount of load they play on the system CPU, cost, licensing models, etc. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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] G729: TC400B vs Software Encoder
Thanks. What do you suggest for system with 15 max simultaneous calls? On Sat, Dec 5, 2009 at 7:49 PM, Kevin P. Fleming wrote: > Jim Boykin wrote: >> Can someone help me decide between TC400B vs Software Encoder. TC400B >> is really expensive and would like to get opinion if it is worth >> considering. Maximum simultaneous call we handle is hardly 10-15. Will >> going to TC400B improves quality. Any other option then TC400B. > > All G.729 encoders produce the same quality, since they all do the same > thing. The only difference between various encoders is the amount of > load they play on the system CPU, cost, licensing models, etc. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- 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
[asterisk-users] G729: TC400B vs Software Encoder
Can someone help me decide between TC400B vs Software Encoder. TC400B is really expensive and would like to get opinion if it is worth considering. Maximum simultaneous call we handle is hardly 10-15. Will going to TC400B improves quality. Any other option then TC400B. Thanks Jim ___ -- 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
