Re: [asterisk-users] Use different local IP for each SIP trunk
Hello, Douglas Mortensen wrote: With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? It's not an asterisk configuration but rather a interface configuration. I need something similar and I use 2 IPs on the same port. In debian, the configuration goes like this: auto eth0 iface eth0 inet static address ip1 netmask netmask1 network network1 broadcast broadcast1 gateway default_gateway auto eth0:0 iface eth0:0 inet static address ip2 netmask netmask2 up route add -net network2 netmask netmask2 gw gw2 And you can add more routes for other specific IPs/networks. José Pablo Méndez Soto wrote: May I ask why do you need different IP addresses to source calls? I mean, its not a common practice, would like to understand the idea behind it. In my case, the operator installed a gateway with a dedicated line and it's connected to the local network, but instead of being 192.168.0.0 it's on 10.0.0.0. So I use this 2 networks in the same NIC in the asterisk machine. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Atxfer for the calling party [SOLVED]
As explained in the posts before, this tread was solved.
Thanks.
On Tue, 2011-12-13 at 17:07 -0200, Antonio Modesto wrote:
On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote:
Hi Antonio,
I'd never had used extensions.ael but in extensions.conf, using
Macro I always set '__TRANSFER_CONTEXT' to the same context of exten
and it works well.
Thanks, it worked, the MACRO_CONTEXT variable was empty, I've set the
value to my extensions context and it worked fine.
Thanks.
2011/12/13 Antonio Modesto mode...@isimples.com.br
Hello everybody,
I found that if i write my macro in the extensions.conf
(not in ael), the atxfer works well, the problem is that ael
uses gosub instead of the Macro() application, which doesn't
change the current context. Does anybody know if i can do
anything to solve this? I know if i rewrite all my macros in
the common way, it will work, but that's a lot of coding for
me.
On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
Nothing?
On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
Hi There,
I'm still having this problem, Does somebody know
what can be happening?
Regards.
On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto
wrote:
Hello,
The exten is the parameter passed to the macro,
which contains the sip device name. I'll change the
name to another less confusing.
* Alexandre, também sou brasileiro hehe, notei que
você já escreveu um livro sobre asterisk, será que
você poderia me ajudar com esse problema? Já tem
alguns dias que estou na luta aqui hehe.
On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller
wrote:
You're using ${exten} inside your macro, you should
use ${EXTEN}.
--
Atenciosamente,
ALEXANDRE KELLER
http://twitter.com/alexandrekeller
http://www.facebook.com/alexandre.keller.BR
Dinheiro é a consequência de um trabalho bem
feito e não o motivo para se fazer um bom trabalho.
P Antes de imprimir pense em seu compromisso com
o Meio Ambiente.
On 11/11/2011, at 08:38, Antonio Modesto wrote:
On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas
wrote:
It can have to do with either the telephones
dial plan or the context in the Asterisk dial
plan combined with your features.conf settings.
I noticed that my problem occurs when i use a
macro to dial sip devices, my dialplan is like
this:
- Each sip device has its own context
- This context includes the outgoing call contexts
that this extension can use for making calls and
includes a context called ramais, which has the
dial plan to call another extensions, it uses a
macro to do this.
Here is the configuration for my extension
modesto :
# sip.conf
[modesto](default_extension)
username=modesto
context=modesto
callerid=modesto 106
callgroup=4
pickupgroup=4
# Default extension template
type=friend
dtmfmode=auto
host=dynamic
disallow=all
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
canreinvite=yes
qualify=no
callcounter=yes
# context for SIP/modesto
context modesto {
includes {
vivo;
tim;
oi;
claro;
vivoddd;
timddd;
oiddd;
claroddd;
embratel;
embratel2;
Re: [asterisk-users] Use different local IP for each SIP trunk
Il 20/12/2011 6.07, Anton Kvashenkin ha scritto: you can add exterin= in sip.conf for each trunk I think this can be used only in [general] section not on peers definition; also useful only when asterisk is behind nat. Not? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] File Convert
Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path: No translator path from g723 to alaw [Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to translate to format g729, source format gsm Even though I have the module format_g729.so. Do I need to have licensed G729 codec for this? or codec_g729.so? Kindly let me know how to convert the file. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Sip Media Call Type
Hi all, I am trying to make a SIP Video and Audio Call, Now when I add at the Asterisk the video Support and the right codec whether I make Audio or Video Call from my clients the Call will be received as Video Call, so the problem is if I make from one client Audio or Video Call it will be recieved as Video Call, Can you plz help me try to solve this problem? Where should I change the Call Media Type at Asterisks Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File Convert
On Tue, Dec 20, 2011 at 05:34:46PM +0530, Gopalakrishnan N wrote: Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path: No translator path from g723 to alaw [Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to translate to format g729, source format gsm Even though I have the module format_g729.so. Do I need to have licensed G729 codec for this? or codec_g729.so? Yes. The g729 codec module requires a per-codec-instance license. In your case you use a single codec for encoding the audio to G.729. BTW: if this is a file you recorded, why convert it from gsm and not from a higher-quality format? If this is from the stanard set of prompts: any chance it is already available as g729? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help_video call not run
Hi all In sip.conf i take as [general] videosupport=yes ; then UDPTL will flow to the remote device [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h261 allow=h263 in extension.conf [employees] exten = 101,1,Dial(SIP/phone1,10) exten = 102,1,Playback(song2_check) in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) i dial 102 from 101 phone 101(xlite) has following codec support for H623 H623+ check log as [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory phone1 goes just hung up. no vedio play I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
Out of curiosity, what is the Polycom script? I obviously haven't moved from 3.2.x firmware yet. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update Did you run your old configurations thru the Polycom script to convert them to work with 3.3+? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind Sent: Friday, December 16, 2011 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update Hello Gord, the line icon is solid black, which should indicate the lines are registered. Marco. On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart gord...@gmail.com wrote: Does the phone show the line as registered? The little phone icon on the display should be solid for a registered line and just a outline for a unregistered line. Using wireshark to watch the SIP traffic is a easy way to ensure the REGISTER signally is complete. On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind marco.mooijek...@gmail.com wrote: Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays SIP URL:... This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
Polycom (r) UC Software: Configuration File Conversion Utility\ On the page http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill Sent: Tuesday, December 20, 2011 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update Out of curiosity, what is the Polycom script? I obviously haven't moved from 3.2.x firmware yet. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update Did you run your old configurations thru the Polycom script to convert them to work with 3.3+? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind Sent: Friday, December 16, 2011 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update Hello Gord, the line icon is solid black, which should indicate the lines are registered. Marco. On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart gord...@gmail.com wrote: Does the phone show the line as registered? The little phone icon on the display should be solid for a registered line and just a outline for a unregistered line. Using wireshark to watch the SIP traffic is a easy way to ensure the REGISTER signally is complete. On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind marco.mooijek...@gmail.com wrote: Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays SIP URL:... This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] India Telecom regulations
Thank you Raj, I hope it will soon require no license as I heard there is a project to change this law, for now I believe I will recommend our office in India to go for license (to bridge to PSTN). Thanks once more for your help! 2011/12/19 Raj Mathur (राज माथुर) r...@linux-delhi.org On Tuesday 20 Dec 2011, khalid touati wrote: Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) If your objective is to originate calls in the US (using whatever technology), route them over SIP and then terminate them to the PSTN in India, then yes: your Indian presence would need a VoIP licence. Similarly for the reverse: originate a call from Indian PSTN to your local office here and route it using VoIP to any destination (whether within India or abroad). A licence is required in that case too. In general, interconnection of two different entities by bridging Indian PSTN with any other technology requires a licence. If you're only doing VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside India then it's permitted in principle. This is why, e.g., Skype is permitted: it doesn't connect to the Indian PSTN at any stage. Once again, IANAL and TINLA. This is purely from my (mostly informed) understanding of the current laws. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] India Telecom regulations
How can we get thise license? Who do we have to pay. Nick. On Tue, Dec 20, 2011 at 9:52 AM, khalid touati khalidtou...@gmail.com wrote: Thank you Raj, I hope it will soon require no license as I heard there is a project to change this law, for now I believe I will recommend our office in India to go for license (to bridge to PSTN). Thanks once more for your help! 2011/12/19 Raj Mathur (राज माथुर) r...@linux-delhi.org On Tuesday 20 Dec 2011, khalid touati wrote: Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) If your objective is to originate calls in the US (using whatever technology), route them over SIP and then terminate them to the PSTN in India, then yes: your Indian presence would need a VoIP licence. Similarly for the reverse: originate a call from Indian PSTN to your local office here and route it using VoIP to any destination (whether within India or abroad). A licence is required in that case too. In general, interconnection of two different entities by bridging Indian PSTN with any other technology requires a licence. If you're only doing VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside India then it's permitted in principle. This is why, e.g., Skype is permitted: it doesn't connect to the Indian PSTN at any stage. Once again, IANAL and TINLA. This is purely from my (mostly informed) understanding of the current laws. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PITCH_SHIFT()
On 20/12/11 01:15 AM, John Jolly wrote: In Leif Madsen's AstriCon 2010 talk titled 5 Things You Didn't Know Asterisk Could Do https://docs.google.com/viewer?a=vq=cache:hCDfIk4pvngJ:leifmadsen.com/sites/default/files/AstriCon%25202010%2520-%25205%2520Things%2520You%2520Didn't%2520Know%2520Asterisk%2520Could%2520Do.pdf+asterisk+dialplan+func+PITCH+SHIFThl=engl=uspid=blsrcid=ADGEEShzSRqJl26lEybK-TvxHL4hKQrd-mBpAapRV6eyI8ST0E5AosCEqp2bm_h5eORZFwwEZDqzEKpT9Fg244nkCgX4BDEGL6bik4Non5_fgm62fzrBxyIXjm1hnqJx2-yGyVlbdXKdsig=AHIEtbQ2NyYajUzeJshmWKAgZEi0RprNjQpli=1 he mentions that the PITCH_SHIFT() function is designed to be used dynamically and can change the pitch of a channel on the fly using features.conf. Can someone provide me with any information of how this would be accomplished for dynamic use? I'm familiar with the dialplan syntax use examples such as: exten = 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave exten = 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more exten = 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch and so forth, but don't understand how these functions would be called dynamically from features.conf. You'd just create the application_map as documented in features.conf and then apply the PITCH_SHIFT() function to whichever channel you want. Untested, but should look something like: pitch_up_them = 3*,peer/both,Set(PITCH_SHIFT(tx)=high) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
Eric Wieling wrote: Polycom (r) UC Software: Configuration File Conversion Utility\ On the pagehttp://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html And for those of us without Windows, this utility appears to work fine under wine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OOH323 config file
Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying to install an H323 link to an Avaya server and the ooh323 module would not load because it could not find its configuration file. The file needs to be renamed. Should this be classified as a bug in the bug tracker? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit # of inbound calls on SIP trunk
Excellent. Do you think these functions would enable me to create rules based on both the concurrent # of inbound and/or outbound calls, or only total # of concurrent calls (agnostic to call direction being inbound vs. outbound)? Thanks, - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 -Original Message- From: Steve Edwards [mailto:asterisk@sedwards.com] Sent: Monday, December 19, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk On Mon, 19 Dec 2011, Douglas Mortensen wrote: I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I’ll need to do some manual dialplan manipulation. The GROUP() and GROUP_COUNT() functions and the GOTOIF() application should do the trick. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GOIP GSM to SIP Gateway?
Hi, Has anyone here any experiencing with linking an Asterisk PBX to a GOIP GSM to SIP Gateway? We've got inbound calls from the GSM network working properly, however, outbound calls seem to randomly choose a SIM line to use. Is there anyway (short of defining dial an 8 from this phone for this trunk to this SIM and a 9 from this phone for a trunk to this SIM) to get it to use certain SIM cards when calls are made from certain phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GOIP GSM to SIP Gateway?
Hello It is possible but how do you have the dialplan ? In your dial plan you can do that Regards On Dec 20, 2011 2:40 PM, Matt mhop...@gmail.com wrote: Hi, Has anyone here any experiencing with linking an Asterisk PBX to a GOIP GSM to SIP Gateway? We've got inbound calls from the GSM network working properly, however, outbound calls seem to randomly choose a SIM line to use. Is there anyway (short of defining dial an 8 from this phone for this trunk to this SIM and a 9 from this phone for a trunk to this SIM) to get it to use certain SIM cards when calls are made from certain phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit # of inbound calls on SIP trunk
Un-top-posting...
On Mon, 19 Dec 2011, Douglas Mortensen wrote:
I have a system with FreePBX, and as far as I can tell it does not
provide a means to limit the number of simultaneous inbound calls on a
SIP trunk. Therefore I suspect that I’ll need to do some manual
dialplan manipulation.
On Mon, 19 Dec 2011, Steve Edwards wrote:
The GROUP() and GROUP_COUNT() functions and the GOTOIF() application
should do the trick.
On Tue, 20 Dec 2011, Douglas Mortensen wrote:
Excellent. Do you think these functions would enable me to create rules
based on both the concurrent # of inbound and/or outbound calls, or only
total # of concurrent calls (agnostic to call direction being inbound
vs. outbound)?
If you want a call to be a member of multiple groups, you have to play
with the category parameter.
exten = *,n,set(GROUP()=incoming)
exten = *,n,set(GROUP(incoming)=no)
exten = *,n,set(GROUP(incoming)=yes)
exten = *,n,set(GROUP()=outgoing)
exten = *,n,set(GROUP(outgoing)=no)
exten = *,n,set(GROUP(outgoing)=yes)
exten = *,n,verbose(incoming count = ${GROUP_COUNT(incoming)})
exten = *,n,verbose(outgoing count = ${GROUP_COUNT(outgoing)})
exten = *,n,verbose(incoming category count =
${GROUP_COUNT(yes@incoming)})
exten = *,n,verbose(outgoing category count =
${GROUP_COUNT(yes@outgoing)})
exten = *,n,verbose(group list is ${GROUP_LIST()})
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] OOH323 config file
On 11-12-20 11:21 AM, Carlos Chavez wrote: Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying to install an H323 link to an Avaya server and the ooh323 module would not load because it could not find its configuration file. The file needs to be renamed. Should this be classified as a bug in the bug tracker? Yes, open an issue on the tracker, if this is the case we can fix it for the next release. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GOIP GSM to SIP Gateway?
On Tue, Dec 20, 2011 at 12:39 PM, Matt mhop...@gmail.com wrote:
Is there anyway (short of defining dial an 8 from this phone for this
trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
get it to use certain SIM cards when calls are made from certain
phones?
You could define multiple contexts with different pattern matches for each
GSM connection and and set your phones to use them, phones 1-3 in context1,
phones 4-6 in context2, etc.
[context1]
_NXX,1,Dial(SIP/GSM1/${EXTEN})
[context2]
_NXX,1,Dial(SIP/GSM2/${EXTEN})
[context3]
_NXX,1,Dial(SIP/GSM3/${EXTEN})
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
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Re: [asterisk-users] GOIP GSM to SIP Gateway?
I would think it would be better to set a variable for each user and then have
a single context with something like:
_NXX,1,Dial(SIP/${WhatToUse}/${EXTEN})
Or something like this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 20, 2011, at 1:03 PM, John Kiniston wrote:
On Tue, Dec 20, 2011 at 12:39 PM, Matt mhop...@gmail.com wrote:
Is there anyway (short of defining dial an 8 from this phone for this
trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
get it to use certain SIM cards when calls are made from certain
phones?
You could define multiple contexts with different pattern matches for each
GSM connection and and set your phones to use them, phones 1-3 in context1,
phones 4-6 in context2, etc.
[context1]
_NXX,1,Dial(SIP/GSM1/${EXTEN})
[context2]
_NXX,1,Dial(SIP/GSM2/${EXTEN})
[context3]
_NXX,1,Dial(SIP/GSM3/${EXTEN})
--
A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build
a wall, set a bone, comfort the dying, take orders, give orders, cooperate,
act alone, solve equations, analyze a new problem, pitch manure, program a
computer, cook a tasty meal, fight efficiently, die gallantly. Specialization
is for insects.
---Heinlein
--
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Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s?
For what it's worth, the phone is getting enough information. The first call works fine - it's the second call that never triggers the pickup screen, though it does cause the lamp to blink for that line. It's like the phone understands ringing but not busy+ringing. I'm tempted to say it's a Polycom firmware issue, but I haven't seen an errata items that matches. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart Sent: Friday, December 16, 2011 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s? It sounds like the phone is not getting enough info to do a directed pickup, have you turned on NotifyCID in sip.conf? If that does'nt work try using the extended BLF stuff (described here http://www.excaliburtech.net/archives/147 and here http://www.voip-info.org/wiki/view/Asterisk+presence) gordu On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill justin.sherr...@americanrocksalt.commailto:justin.sherr...@americanrocksalt.com wrote: This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825tel:585-991-6825 F: 585-991-6925tel:585-991-6925 C: 585-298-6826tel:585-298-6826 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit # of inbound calls on SIP trunk
Well freepbx has that in the gui you should read the tool tips
Read the trunk limit tooltip
-Original Message-
From: Steve Edwards asterisk@sedwards.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 20 Dec 2011 12:16:48
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk
Un-top-posting...
On Mon, 19 Dec 2011, Douglas Mortensen wrote:
I have a system with FreePBX, and as far as I can tell it does not
provide a means to limit the number of simultaneous inbound calls on a
SIP trunk. Therefore I suspect that I’ll need to do some manual
dialplan manipulation.
On Mon, 19 Dec 2011, Steve Edwards wrote:
The GROUP() and GROUP_COUNT() functions and the GOTOIF() application
should do the trick.
On Tue, 20 Dec 2011, Douglas Mortensen wrote:
Excellent. Do you think these functions would enable me to create rules
based on both the concurrent # of inbound and/or outbound calls, or only
total # of concurrent calls (agnostic to call direction being inbound
vs. outbound)?
If you want a call to be a member of multiple groups, you have to play
with the category parameter.
exten = *,n,set(GROUP()=incoming)
exten = *,n,set(GROUP(incoming)=no)
exten = *,n,set(GROUP(incoming)=yes)
exten = *,n,set(GROUP()=outgoing)
exten = *,n,set(GROUP(outgoing)=no)
exten = *,n,set(GROUP(outgoing)=yes)
exten = *,n,verbose(incoming count = ${GROUP_COUNT(incoming)})
exten = *,n,verbose(outgoing count = ${GROUP_COUNT(outgoing)})
exten = *,n,verbose(incoming category count =
${GROUP_COUNT(yes@incoming)})
exten = *,n,verbose(outgoing category count =
${GROUP_COUNT(yes@outgoing)})
exten = *,n,verbose(group list is ${GROUP_LIST()})
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Populate CDR issues
Hi Mike,
I've tried updating my CDR's via the h exten but with no success. I've tried
with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the
value refused to appear in my CDR (even though I see the Set() application
being executed in the console under the h exten). Thank you for your suggestion
though...
Any other thoughts are welcome.
Kind Regards,
Harel Cohen
-Original Message-
Date: Mon, 12 Dec 2011 13:41:31 -0700
From: Mike Diehl mdi...@diehlnet.com
Subject: Re: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
Cc: Harel Cohen ha...@easycall.gi
Message-ID: 201112121341.32142.mdi...@diehlnet.com
Content-Type: Text/Plain; charset=iso-8859-1
On Monday 12 December 2011 4:28:17 am Harel Cohen wrote:
Danny,
Why would you think this is a circumvent? I'm using a nice feature
of 1.8 where I can create any CDR field I like and populate it by
using the
CDR(fieldname) function. While all other fields that I created are
populated properly (however before the 'dial' commences) it seems like
at this point of the dial plan the CDR is closed for editing even
though I configured endbeforehexten=no in my cdr.conf.
I agree, this is a perfectly valid use of the CDR. I do the same thing, btw.
I think what you are seeing is that when your call starts, Asterisk creates a
record, either in memory, or in a db transaction. When the call is torn down,
the record is updated and committed to the db. The down-shot is that any
changes you make to the db record get clobbered by this last update.
I ended up making some of my updates in the hang-up phase via the h
extension. See if that will do what you need.
--
Take care and have fun,
Mike Diehl.
--
Date: Thu, 1 Dec 2011 12:57:56 +0100
From: Harel Cohen ha...@easycall.gi
Subject: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Message-ID:
CABC006234837141A279831F68041406CABE9415A4@ECS.EasyCall.local
Content-Type: text/plain; charset=us-ascii
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015, SIP/office_ServerRoom,,FgU(jump2SetVar))
in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016, CDR(chanoutsigip)=192.168.20.226:5065)
in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[asterisk-users] sendvoicemail=yes not quite working
I have a system working great with the exception of the sendvoicemail=yes voicemail.conf option. I can not figure out what I am missing or have configured wrong... While in voicemail after selecting 3 for advanced options, then 5 to leave a message I am directed to the correct mailbox. But after hearing the mailbox number/name announcement I am immediately taken back to my mailbox. No option is given to leave a message. I can forward messages, but can't leave a message. All other aspects of the voicemail system I have tested work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with about 120 phones. Here is the vebose/debug output of that part of the call. == Using SIP RTP CoS mark 5 -- Executing [8000@LocalSets:1] VoiceMailMain(SIP/3323-0499, ) in new stack -- Playing 'vm-login.ulaw' (language 'en') -- Playing 'vm-password.ulaw' (language 'en') -- Playing 'vm-youhave.ulaw' (language 'en') -- Playing 'vm-no.ulaw' (language 'en') -- Playing 'vm-messages.ulaw' (language 'en') -- Playing 'vm-leavemsg.ulaw' (language 'en') -- Playing 'vm-starmain.ulaw' (language 'en') -- Playing 'vm-extension.ulaw' (language 'en') -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin' (language 'en') -THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX == Using SIP RTP CoS mark 5 -- Playing 'vm-opts.ulaw' (language 'en') == Spawn extension (LocalSets, 8000, 1) exited non-zero on 'SIP/3323-0499' Thanks! Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendvoicemail=yes not quite working
On Tue, Dec 20, 2011 at 8:53 PM, Todd Routhier fonema...@gmail.com wrote: On Tue, Dec 20, 2011 at 7:03 PM, M Maki mma...@verizon.net wrote: I have a system working great with the exception of the sendvoicemail=yes voicemail.conf option. I can not figure out what I am missing or have configured wrong... While in voicemail after selecting 3 for advanced options, then 5 to leave a message I am directed to the correct mailbox. But after hearing the mailbox number/name announcement I am immediately taken back to my mailbox. No option is given to leave a message. I can forward messages, but can't leave a message. All other aspects of the voicemail system I have tested work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with about 120 phones. Here is the vebose/debug output of that part of the call. == Using SIP RTP CoS mark 5 -- Executing [8000@LocalSets:1] VoiceMailMain(SIP/3323-0499, ) in new stack -- Playing 'vm-login.ulaw' (language 'en') -- Playing 'vm-password.ulaw' (language 'en') -- Playing 'vm-youhave.ulaw' (language 'en') -- Playing 'vm-no.ulaw' (language 'en') -- Playing 'vm-messages.ulaw' (language 'en') -- Playing 'vm-leavemsg.ulaw' (language 'en') -- Playing 'vm-starmain.ulaw' (language 'en') -- Playing 'vm-extension.ulaw' (language 'en') -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin' (language 'en') -THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX == Using SIP RTP CoS mark 5 -- Playing 'vm-opts.ulaw' (language 'en') == Spawn extension (LocalSets, 8000, 1) exited non-zero on 'SIP/3323-0499' Thanks! Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Pretty sure you need this in your voicemail.conf file: sendvoicemail=yes sendvoicemailThis setting takes a *yes* or *no* value. It enables the Leave a message menu option from the Advanced Options menu which allows the voicemail user to send a message to another voicemail user. Oh, wow.. Nevermind, you started your original post saying you have that option set. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendvoicemail=yes not quite working
On Tue, Dec 20, 2011 at 7:03 PM, M Maki mma...@verizon.net wrote: I have a system working great with the exception of the sendvoicemail=yes voicemail.conf option. I can not figure out what I am missing or have configured wrong... While in voicemail after selecting 3 for advanced options, then 5 to leave a message I am directed to the correct mailbox. But after hearing the mailbox number/name announcement I am immediately taken back to my mailbox. No option is given to leave a message. I can forward messages, but can't leave a message. All other aspects of the voicemail system I have tested work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with about 120 phones. Here is the vebose/debug output of that part of the call. == Using SIP RTP CoS mark 5 -- Executing [8000@LocalSets:1] VoiceMailMain(SIP/3323-0499, ) in new stack -- Playing 'vm-login.ulaw' (language 'en') -- Playing 'vm-password.ulaw' (language 'en') -- Playing 'vm-youhave.ulaw' (language 'en') -- Playing 'vm-no.ulaw' (language 'en') -- Playing 'vm-messages.ulaw' (language 'en') -- Playing 'vm-leavemsg.ulaw' (language 'en') -- Playing 'vm-starmain.ulaw' (language 'en') -- Playing 'vm-extension.ulaw' (language 'en') -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin' (language 'en') -THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX == Using SIP RTP CoS mark 5 -- Playing 'vm-opts.ulaw' (language 'en') == Spawn extension (LocalSets, 8000, 1) exited non-zero on 'SIP/3323-0499' Thanks! Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Pretty sure you need this in your voicemail.conf file: sendvoicemail=yes sendvoicemailThis setting takes a *yes* or *no* value. It enables the Leave a message menu option from the Advanced Options menu which allows the voicemail user to send a message to another voicemail user. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use different local IP for each SIP trunk
Externip support per device in sip.conf http://edvina.net/products/edvx/ 2011/12/20 giovanni.v i...@keybits.org Il 20/12/2011 6.07, Anton Kvashenkin ha scritto: you can add exterin= in sip.conf for each trunk I think this can be used only in [general] section not on peers definition; also useful only when asterisk is behind nat. Not? -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help_video call not run
Hi all In sip.conf i take as [general] videosupport=yes [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h261 allow=h263 in extension.conf [employees] exten = 101,1,Dial(SIP/phone1,10) exten = 102,1,Playback(song2_check) in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) i dial 102 from 101 phone 101(xlite) has following codec support for H623 H623+ check log as [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory phone1 goes just hung up. no vedio play I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue not skipping ringing phone
I have a queue that distributes calls among 3 phones. When a phone is in use (including on hold), queue skips that device and sends the call to the next available one as expected. On the other hand, if a call comes in while one of the phones is ringing, the queue doesn't seem to recognize that phone as in use and sends the second call to the ringing phone. If the first call is answered, the second call is sent to the next available phone right away. I'm new to asterisk and wondering if this is normal; I thought the ringing phone would be skipped as in use as well. Is there a setting on the asterisk side that I can use to force the queue to skip the ringing phone, or should this somehow be done on the phone itself? Thanks, Matt Below is the queues.conf: [qtemplate] announce-frequency=0 announce-holdtime=no announce-position=no autofill=yes eventmemberstatus=no eventwhencalled=no joinempty=strict leavewhenempty=strict maxlen=0 memberdelay=0 penaltymemberslimit=0 periodic-announce-frequency=0 queue-callswaiting=silence/1 Sendqueue-thereare=silence/1 queue-youarenext=silence/1 reportholdtime=no ringinuse=no servicelevel=60 strategy=rrmemory timeout=0 timeoutpriority=app timeoutrestart=no retry=0 weight=0 wrapuptime=0 musicclass=default monitor-type=MixMonitor monitor-format=wav [q1000](qtemplate) member=Local/1001@handle-queue,,,SIP/1001 member=Local/1002@handle-queue,,,SIP/1002 member=Local/1003@handle-queue,,,SIP/1003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help_video call not run
Hi what is the format of the file you are trying to play with exact codec info. On Tue, Dec 20, 2011 at 19:17, Durgesh Mishra durgesh.mis...@rancoretech.com wrote: Hi all In sip.conf i take as [general] videosupport=yes ; then UDPTL will flow to the remote device [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h261 allow=h263 in extension.conf [employees] exten = 101,1,Dial(SIP/phone1,10) exten = 102,1,Playback(song2_check) in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) i dial 102 from 101 phone 101(xlite) has following codec support for H623 H623+ check log as [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory phone1 goes just hung up. no vedio play I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Which switch to play with LLDP-MED
Hi, I would like to play with LLDP-MED in a lab, and specifically, to test phone provisionning and auto-configuration (assign phones to VLANs, ...). Eight 10/100 PoE ports would be enough for me. Which model would you recommend ? Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help_video call not run
- Forwarded Message - From: Durgesh Mishra durgesh.mis...@rancoretech.com To: asterisk-users asterisk-users@lists.digium.com Sent: Wednesday, December 21, 2011 10:36:06 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Help_video call not run Hi all In sip.conf i take as [general] videosupport=yes [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h261 allow=h263 in extension.conf [employees] exten = 101,1,Dial(SIP/phone1,10) exten = 102,1,Playback(song2_check) in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) i check in /var/lib/asterisk/sounds/en its shows as -rw-r--r-- 1 root root 89726486 Dec 20 10:18 song_check.mp4 i check codecs of file through VLC player , its shows as its codec is using is H264. i check through cli core show codecs : 131072 (1 17) (0x2) image png (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) 2097152 (1 21) (0x20) video h264 (H.264 Video) 4194304 (1 22) (0x40) video mpeg4 (MPEG4 Video) ::: i dial 102 from 101 phone 101(xlite) has following codec support for H623 H623+ check log as [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory phone1 goes just hung up. no vedio play I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
