Re: [asterisk-users] Recommended *WRT router to run Asterisk?
I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. Hi, Can you add a DSL modem inside these boxes ? Cheers -- _ -- 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] Recommended *WRT router to run Asterisk?
On Thu, 18 Nov 2010, John Novack wrote: Not really in production But for a SIP/IAX Asterisk box, it works! there is a Dockstar hacking site that de-nuts the boot code and allows booting from a 1-2 gig flash ( I have not had good luck with 4 and 8 gig flash, but it could be the flash sticks. Loading Debian squeeze onto the flash, configure Debian not to use the swap, then wget and compile Asterisk. the make file needs to be modified to specify arm5 rather than the longer name configure generates. adding some additional packages to the Debian load will be needed. Lenny also works. the Dockstar only has 128M of ram. the more expensive Sheeva I believe has more, but for a small office or home it just sits there and works! remember to noload all the unwanted modules as well. I am no Linux eggspurt, but I got this working, with the help of Google, in a few hours. Sometimes smaller is better, or at least it can be fun! Interestingly for commercial units, I've had the opposite experience - I've found that my (business) customers just will not pay for something tiny that's capable of supporting 30 phones... I did have a look at the GuruPlug stuff recently, but it's just not going to be sensible for me to put any effort into it as people won't buy it. Even my smallest box at 150mm square is verging on the unbelievable for some people - especially those used to a PBX taking up a whole rack, or having 2 or more large units bolted to a wall... Still, for home/hobby use these little things are great! Gordon -- _ -- 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] Recommended *WRT router to run Asterisk?
Hi, No we can not add a DSL modem.But i saw on the net that this works for call forward feature on TDM400 FXS card.So i tried it.The data is entering the CFIM family in the database which can be seen but only the call is not forwarded Regards, Aparna On Fri, Nov 19, 2010 at 2:56 PM, Olivier oza_4...@yahoo.fr wrote: I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. Hi, Can you add a DSL modem inside these boxes ? Cheers -- _ -- 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] Using Local Asterisk Server with Siphon - Can't hear voice issue
Hi, It worked finally with GSM Codec only enabled at client side.. Initially with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All enabled by setting [CLI] sip set debug on I saw asterisk having following logs.. -- Remotely bridging SIP/macbook-0041 and SIP/tharindu-0042 set_destination: Parsing sip:thari...@192.168.1.3sip%3athari...@192.168.1.3 : 64540;ob for address/port to send to set_destination: set destination to 192.168.1.3:64540 Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP -- So I enabled GSM only .. then everything got solved.. :) --- I would like to register SIP users to the server using kind of web service.. instead of manually entering extensions and users using configuration files.. TO achieve this could some body point some instructions. ?? One more thing.. Is it possible to automatically reload the servers after some small time ?? instead of manually typing the command on [CLI] console. ?? Thanks and Kind Regards, Tharindu Madushanka, tharindufit.wordpress.com -- _ -- 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] Ekiga can register but not my IP phone
Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with the same SIP account. Here is a part of my sip.conf : [general] dtmfmode=auto language=fr ; pour les messages lus par asterisk disallow=all allow=ulaw allow=alaw allow=speex [siemens] type=friend context=interne host=dynamic secret= When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some information. It seems that asterisk receives the rengistration request but doesn't answer to it. Here are the logs : http://server.chab.info/Registration_logs_ip_phone.txt Using Ekiga with the same SIP account (name is siemens) and from the same physical location works well : http://server.chab.info/Registration_logs_ekiga.txt I didn't change anything about asterisk config (except sip.conf and extensions.conf). If you have any idea, please share it with me, i really don't to do to fix this problem... Thanks in advance ! -- _ -- 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] Ekiga can register but not my IP phone
On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). I am assuming you mean Asterisk on Ubuntu inside the LAN But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with the same SIP account. Is this outside the LAN? Is there NAT in between? SIP is a pain in the ass with NAT, so it's the only thing I can think of. Usually in my experience it's the other way around! Ekiga is the one that doesn't work and tends to be very quirky (takes a long time to quit, has strange registration quirks, etc.), I mean when compared to HW SIP device. Here is a part of my sip.conf : [general] dtmfmode=auto language=fr ; pour les messages lus par asterisk disallow=all allow=ulaw allow=alaw allow=speex [siemens] type=friend context=interne host=dynamic secret= When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some information. It seems that asterisk receives the rengistration request but doesn't answer to it. Here are the logs : http://server.chab.info/Registration_logs_ip_phone.txt Using Ekiga with the same SIP account (name is siemens) and from the same physical location works well : http://server.chab.info/Registration_logs_ekiga.txt I didn't change anything about asterisk config (except sip.conf and extensions.conf). If you have any idea, please share it with me, i really don't to do to fix this problem... Thanks in advance ! The only thing I can think of are NAT issues with SIP. If you are in fact NATing try the Siemens phone to a direct IP to the server (no NAT, firewall, etc.) and see. -- Alejandro Imass -- _ -- 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] Ekiga can register but not my IP phone
Thanks Alejandro, you were right it was just a NAT problem ! i add a stun server in the phone configuration and it works :) 2010/11/19, Alejandro Imass a...@p2ee.org: On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). I am assuming you mean Asterisk on Ubuntu inside the LAN But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with the same SIP account. Is this outside the LAN? Is there NAT in between? SIP is a pain in the ass with NAT, so it's the only thing I can think of. Usually in my experience it's the other way around! Ekiga is the one that doesn't work and tends to be very quirky (takes a long time to quit, has strange registration quirks, etc.), I mean when compared to HW SIP device. Here is a part of my sip.conf : [general] dtmfmode=auto language=fr ; pour les messages lus par asterisk disallow=all allow=ulaw allow=alaw allow=speex [siemens] type=friend context=interne host=dynamic secret= When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some information. It seems that asterisk receives the rengistration request but doesn't answer to it. Here are the logs : http://server.chab.info/Registration_logs_ip_phone.txt Using Ekiga with the same SIP account (name is siemens) and from the same physical location works well : http://server.chab.info/Registration_logs_ekiga.txt I didn't change anything about asterisk config (except sip.conf and extensions.conf). If you have any idea, please share it with me, i really don't to do to fix this problem... Thanks in advance ! The only thing I can think of are NAT issues with SIP. If you are in fact NATing try the Siemens phone to a direct IP to the server (no NAT, firewall, etc.) and see. -- Alejandro Imass -- _ -- 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] Ekiga can register but not my IP phone
On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier c...@chab.info wrote: Thanks Alejandro, you were right it was just a NAT problem ! i add a stun server in the phone configuration and it works :) Cool. Also Asterisk SIP conf file has some NAT settings as well that you can play with and perhaps do away with the stun server config in the phone. Here is a great article that explains in detail the issues with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html 2010/11/19, Alejandro Imass a...@p2ee.org: On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with -- _ -- 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 1.8 and Dial(SIP/peer_name) to undefined peer
Hi, In Asterisk 1.8.0 dialplan command Dial(SIP/peer_name) produces errors if no such peer_name defined instead of just saying peer not found: [Nov 19 20:01:23] ERROR[7827]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo(sdf, (null), ...): Name or service not known [Nov 19 20:01:23] WARNING[7827]: chan_sip.c:5041 create_addr: No such host: sdf [Nov 19 20:01:23] NOTICE[7827]: channel.c:5106 __ast_request_and_dial: Unable to request channel SIP/sdf I didn't find any bug report regarding this issue. Is there any setting in sip.conf to disable host resolving in case of undefined peer name? -- Best regards, Grigoriy Puzankin -- _ -- 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] How to append custom option to Contact: header on outgoing SIP INVITE msgs?
On Fri, Nov 5, 2010 at 10:58 AM, Bob Beers bob.be...@gmail.com wrote:
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: bob:3125551...@10.10.10.10;SITE-ID=us.here
where SITE-ID=us.here is set in a config file that * parses on
startup. Or in a Dial() command option? Or I don't care exactly
how. :-)
It is possible to affect the Contact: header via a line in sip.conf:
register = toronto:welc...@192.168.1.101/contact
but I can't get it to also accept any ;X=Y params for the
contact.
I can add a custom Contact header in the dialplan with SipAddHeader,
but then I have two. SipRemoveHeader only removes headers
previously added by SipAddHeader, so no luck there.
I have googled, and searched the asterisk-users mailing list archives
and not yet found a solution. I did see some work back in 2004
(issues 732 and 777) which mentioned not stripping contact header
parameters from arriving requests/registrations, but nothing about
creating any such parameters.
Thanks for any help/hints,
Am I on the wrong list?
I have not noticed any replies, so I have moved forward with this idea:
# cat redhat/SOURCES/asterisk-1.8.0-beta2-Contactoption-bbeers03.patch
--- asterisk-1.8.0-beta2.orig/channels/chan_sip.c 2010-07-26
15:59:03.0 -0400
+++ asterisk-1.8.0-beta2/channels/chan_sip.c2010-11-05
12:18:53.0 -0400
@@ -722,6 +755,7 @@ static unsigned int global_cos_video;
static unsigned int global_cos_text; /*! 802.1p class of service
for text RTP packets */
static unsigned int recordhistory; /*! Record SIP history. Off
by default */
static unsigned int dumphistory; /*! Dump history to verbose
before destroying SIP dialog */
+static char global_contactoption[AST_MAX_EXTENSION];/*! string
to append to Contact: for the SIP channel */
static char global_useragent[AST_MAX_EXTENSION];/*! Useragent
for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION]; /*! SDP session
name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*! SDP owner
name for the SIP channel */
@@ -10936,12 +12018,14 @@ static void extract_uri(struct sip_pvt *
static void build_contact(struct sip_pvt *p)
{
if (p-socket.type == SIP_TRANSPORT_UDP) {
- ast_string_field_build(p, our_contact, sip:%s%s%s, p-exten,
- ast_strlen_zero(p-exten) ? : @,
ast_sockaddr_stringify(p-ourip));
+ ast_string_field_build(p, our_contact,
sip:%s%s%s%s%s, p-exten,
+ ast_strlen_zero(p-exten) ? : @,
ast_sockaddr_stringify(p-ourip),
+ ast_strlen_zero(global_contactoption) ? :
;, global_contactoption);
} else {
- ast_string_field_build(p, our_contact,
sip:%s%s%s;transport=%s, p-exten,
+ ast_string_field_build(p, our_contact,
sip:%s%s%s;transport=%s%s%s, p-exten,
ast_strlen_zero(p-exten) ? : @,
ast_sockaddr_stringify(p-ourip),
- get_transport(p-socket.type));
+ get_transport(p-socket.type),
+ ast_strlen_zero(global_contactoption) ? :
;, global_contactoption);
}
}
@@ -26403,6 +28341,8 @@ static int reload_config(enum channelrel
global_relaxdtmf = ast_true(v-value);
} else if (!strcasecmp(v-name, vmexten)) {
ast_copy_string(default_vmexten, v-value,
sizeof(default_vmexten));
+ } else if (!strcasecmp(v-name, contactoption)) {
+ ast_copy_string(global_contactoption,
v-value, sizeof(global_contactoption));
} else if (!strcasecmp(v-name, rtptimeout)) {
if ((sscanf(v-value, %30d,
global_rtptimeout) != 1) || (global_rtptimeout 0)) {
ast_log(LOG_WARNING, '%s' is not a
valid RTP hold time at line %d. Using default.\n, v-value,
v-lineno);
Then I add to the [general] section of sip.conf,
contactoption=SITE-ID=us.here
and it works for me, but I still wonder if there is a better way.
--
Thanks,
-Bob Beers
--
_
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Re: [asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?
The linewrapping by gmail of the patch file makes it difficult to read.
So, I added it as an attachment for any interested readers.
--
-Bob
--- asterisk-1.8.0-beta2.orig/channels/chan_sip.c 2010-07-26 15:59:03.0 -0400
+++ asterisk-1.8.0-beta2/channels/chan_sip.c 2010-11-05 12:18:53.0 -0400
@@ -722,6 +755,7 @@ static unsigned int global_cos_video;
static unsigned int global_cos_text; /*! 802.1p class of service for text RTP packets */
static unsigned int recordhistory; /*! Record SIP history. Off by default */
static unsigned int dumphistory; /*! Dump history to verbose before destroying SIP dialog */
+static char global_contactoption[AST_MAX_EXTENSION];/*! string to append to Contact: for the SIP channel */
static char global_useragent[AST_MAX_EXTENSION];/*! Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION]; /*! SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*! SDP owner name for the SIP channel */
@@ -10936,12 +12018,14 @@ static void extract_uri(struct sip_pvt *
static void build_contact(struct sip_pvt *p)
{
if (p-socket.type == SIP_TRANSPORT_UDP) {
- ast_string_field_build(p, our_contact, sip:%s%s%s, p-exten,
- ast_strlen_zero(p-exten) ? : @, ast_sockaddr_stringify(p-ourip));
+ ast_string_field_build(p, our_contact, sip:%s%s%s%s%s, p-exten,
+ ast_strlen_zero(p-exten) ? : @, ast_sockaddr_stringify(p-ourip),
+ ast_strlen_zero(global_contactoption) ? : ;, global_contactoption);
} else {
- ast_string_field_build(p, our_contact, sip:%s%s%s;transport=%s, p-exten,
+ ast_string_field_build(p, our_contact, sip:%s%s%s;transport=%s%s%s, p-exten,
ast_strlen_zero(p-exten) ? : @, ast_sockaddr_stringify(p-ourip),
- get_transport(p-socket.type));
+ get_transport(p-socket.type),
+ ast_strlen_zero(global_contactoption) ? : ;, global_contactoption);
}
}
@@ -26403,6 +28341,8 @@ static int reload_config(enum channelrel
global_relaxdtmf = ast_true(v-value);
} else if (!strcasecmp(v-name, vmexten)) {
ast_copy_string(default_vmexten, v-value, sizeof(default_vmexten));
+ } else if (!strcasecmp(v-name, contactoption)) {
+ ast_copy_string(global_contactoption, v-value, sizeof(global_contactoption));
} else if (!strcasecmp(v-name, rtptimeout)) {
if ((sscanf(v-value, %30d, global_rtptimeout) != 1) || (global_rtptimeout 0)) {
ast_log(LOG_WARNING, '%s' is not a valid RTP hold time at line %d. Using default.\n, v-value, v-lineno);
--
_
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Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem
Why not just use tiff2pdf ? tiff2pdf input.tif -o output.pdf William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Michael Sent: Friday, November 19, 2010 9:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error FAX handle 0: failed to queue document 'filename.tif', so we set it to 1680x2285, but it's still rejected. Is there a way to debug this further and fix it? We often have tif source files that we prefer to send, without converting to pdf and back to tif. Thank you in advance, Michael -- _ -- 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] Recommended *WRT router to run Asterisk?
On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote: Interestingly for commercial units, I've had the opposite experience - I've found that my (business) customers just will not pay for something tiny that's capable of supporting 30 phones... I did have a look at the GuruPlug stuff recently, but it's just not going to be sensible for me to put any effort into it as people won't buy it. Even my smallest box at 150mm square is verging on the unbelievable for some people - especially those used to a PBX taking up a whole rack, or having 2 or more large units bolted to a wall... Still, for home/hobby use these little things are great! It seems to me that there's definitely a break point below which very small hardware platforms are really only suitable for hobby or very niche applications. I once ran Astlinux on a Gumstix board just to see if it was possible. But beyond embedded applications it actually creates problems to have such small hardware. OTOH, net-tops like the the Fit-PC2i are really very interesting, and servicable in small producti roles. http://www.mgraves.org/2010/07/d-i-y-asterisk-appliances-a-question-of-s cale/ Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] FFA (Fax For Asterisk) tif file (size) problem
So tiff2pdf and then gs back to tif? I was hoping for a cleaner method Original Message Why not just use tiff2pdf ? tiff2pdf input.tif -o output.pdf William Stillwell -- _ -- 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] Recommended *WRT router to run Asterisk?
On 11/19/2010 10:10 AM, Michael Graves wrote: On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote: Interestingly for commercial units, I've had the opposite experience - I've found that my (business) customers just will not pay for something tiny that's capable of supporting 30 phones... I did have a look at the GuruPlug stuff recently, but it's just not going to be sensible for me to put any effort into it as people won't buy it. Even my smallest box at 150mm square is verging on the unbelievable for some people - especially those used to a PBX taking up a whole rack, or having 2 or more large units bolted to a wall... Still, for home/hobby use these little things are great! It seems to me that there's definitely a break point below which very small hardware platforms are really only suitable for hobby or very niche applications. I once ran Astlinux on a Gumstix board just to see if it was possible. But beyond embedded applications it actually creates problems to have such small hardware. OTOH, net-tops like the the Fit-PC2i are really very interesting, and servicable in small producti roles. http://www.mgraves.org/2010/07/d-i-y-asterisk-appliances-a-question-of-s cale/ The problem is people want to find the $50 embedded solution and then use it where the $1000 solution is really needed. What is nice is when the $50 hardware and the $1000 hardware run exactly the same software so other than the drivers for the hardware itself, everything else behaves the same way and its easy to move around configurations to grow. (I am not talking about asterisk specifically, just generally about routers, backup devices, media servers, etc) Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] FFA (Fax For Asterisk) tif file (size) problem
On Fri, Nov 19, 2010 at 9:42 AM, Michael voip.quest...@gmail.com wrote: Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error FAX handle 0: failed to queue document 'filename.tif', so we set it to 1680x2285, but it's still rejected. Is there a way to debug this further and fix it? We often have tif source files that we prefer to send, without converting to pdf and back to tif. Thank you in advance, Michael I don't know if this is the case or not, but check for differences between the two tiff files. I wonder if one is compressed and the other is not? -M -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call from Alice, transparently. Tried to set the callerid but Charlie sees my telco line number, not the callerid of Alice. How can I do this? Thank you. Giorgio -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, November 19, 2010 9:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call from Alice, transparently. Tried to set the callerid but Charlie sees my telco line number, not the callerid of Alice. How can I do this? Thank you. Giorgio -- We know that Alice and Charlie are both on external trunks. We DON'T know what flavor of Asterisk you are using, but it probably doesn't matter your call is going like this ID #1 -- asterisk -- destination. If destination were internal, ID#1 would remain intact, but since you are opening a new trunk to forward the call, you lose ID#2 and replace it with your Telco ID. You could spoof this depending on your asterisk version/telco arrangement, but by default, things are as you described. -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Please try this in your dialplan Set(CALLERID(name)=${CALLERID(num)})
Some where I tried and it worked with VoIP account A to B as VoIP trunk and
B forward the call to C whereas in C A's number will be displayed.
If you could paste more details as Danny said that would help the list to
assist you more.
On Fri, Nov 19, 2010 at 9:11 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring call
from
ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has transferred the
call as if Charlie is receiving the call from Alice, transparently.
Tried to set the callerid but Charlie sees my telco line number, not the
callerid of Alice.
How can I do this?
Thank you.
Giorgio
--
We know that Alice and Charlie are both on external trunks. We DON'T know
what flavor of Asterisk you are using, but it probably doesn't matter your
call is going like this
ID #1 -- asterisk -- destination.
If destination were internal, ID#1 would remain intact, but since you are
opening a new trunk to forward the call, you lose ID#2 and replace it with
your Telco ID. You could spoof this depending on your asterisk
version/telco arrangement, but by default, things are as you described.
--
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VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com
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Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id checkbox while creating VoIP trunk then it started
working for me..
On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Please try this in your dialplan Set(CALLERID(name)=${CALLERID(num)})
Some where I tried and it worked with VoIP account A to B as VoIP trunk and
B forward the call to C whereas in C A's number will be displayed.
If you could paste more details as Danny said that would help the list to
assist you more.
On Fri, Nov 19, 2010 at 9:11 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring call
from
ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has transferred the
call as if Charlie is receiving the call from Alice, transparently.
Tried to set the callerid but Charlie sees my telco line number, not the
callerid of Alice.
How can I do this?
Thank you.
Giorgio
--
We know that Alice and Charlie are both on external trunks. We DON'T know
what flavor of Asterisk you are using, but it probably doesn't matter your
call is going like this
ID #1 -- asterisk -- destination.
If destination were internal, ID#1 would remain intact, but since you are
opening a new trunk to forward the call, you lose ID#2 and replace it with
your Telco ID. You could spoof this depending on your asterisk
version/telco arrangement, but by default, things are as you described.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com
--
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Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, November 19, 2010 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk Hi Danny, I'm using Asterisk 1.4 and I'm using SetCallerPres and Set(CALLERID(name)=XX) apps but I always get my telco callerid. Which Asterisk version would you suggest? Thanks! Giorgio I don't think the version is relevant to this query. I would see what followme might do for you in this case. -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Hi Gopalakrishnan A.N,
I tried it but it seems like my telco is overwriting the value I set as
callerid.
Maybe it is possible with Voip providers only.
Giorgio Incantalupo
Gopalakrishnan A.N wrote:
Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id checkbox while creating VoIP trunk then it
started working for me..
On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N sai...@gmail.com
mailto:sai...@gmail.com wrote:
Please try this in your dialplan
Set(CALLERID(name)=${CALLERID(num)})
Some where I tried and it worked with VoIP account A to B as VoIP
trunk and B forward the call to C whereas in C A's number will be
displayed.
If you could paste more details as Danny said that would help the
list to assist you more.
On Fri, Nov 19, 2010 at 9:11 PM, Danny Nicholas da...@debsinc.com
mailto:da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when
transferring call from
ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has
transferred the
call as if Charlie is receiving the call from Alice,
transparently.
Tried to set the callerid but Charlie sees my telco line
number, not the
callerid of Alice.
How can I do this?
Thank you.
Giorgio
--
We know that Alice and Charlie are both on external trunks.
We DON'T know
what flavor of Asterisk you are using, but it probably doesn't
matter your
call is going like this
ID #1 -- asterisk -- destination.
If destination were internal, ID#1 would remain intact, but
since you are
opening a new trunk to forward the call, you lose ID#2 and
replace it with
your Telco ID. You could spoof this depending on your asterisk
version/telco arrangement, but by default, things are as you
described.
--
_
-- 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
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
mailto:sip%3asai...@gtalk2voip.com
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com mailto:sip%3asai...@gtalk2voip.com
--
_
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Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Hi Danny, I'm using Asterisk 1.4 and I'm using SetCallerPres and Set(CALLERID(name)=XX) apps but I always get my telco callerid. Which Asterisk version would you suggest? Thanks! Giorgio Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, November 19, 2010 9:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call from Alice, transparently. Tried to set the callerid but Charlie sees my telco line number, not the callerid of Alice. How can I do this? Thank you. Giorgio -- _ -- 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 with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and canreinvite=no. I want the rtp traffic goes through asterisk. I can reproduce the waring message below when a peer uses a different codec that ulaw. when the UA send out a number to route through the trunk, the warning message is not displayed initially, but only when the ring tone starts, then the mesage appears repeatedly and stops when the called peer answers, and the call is bridged successfully without problems. Im using asterisk 1.8.0 from sip.conf [Trunk-out] type=peer host=*.*.*.* context=from-Trunk-out insecure=port,invite qualify=yes disallow=all allow=ulaw directmedia=no canreinvite=no dtmfmode=rfc2833 [ua1] type=friend secret=* host=dynamic nat=yes qualify=6 disallow=all allow=ulaw allow=gsm allow=ilbc allow=alaw qualifyfreq=9 context=sswtrunks directmedia=no canreinvite=no [ua2] ... == Using SIP RTP CoS mark 5 -- Executing [...@sswtrunks:1] Dial(SIP/ua1-0776, SIP/Trunk-out/***,180,tTr) in new stack == Using SIP RTP CoS mark 5 -- Called Trunk-out/** -- SIP/Trunk-out-0777 is ringing -- SIP/Trunk-out-0777 is making progress passing it to SIP/ua1-0776 [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) [Nov 19 17:08:37] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) over and over [Nov 19 17:08:38] WARNING[8377]: chan_sip.c:6027 sip_write: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm) -- SIP/Trunk-out-0777 answered SIP/ua1-0776 == Spawn extension (sswtrunks, , 1) exited non-zero on 'SIP/ua1-0776' any idea ?? thanks in advance -- _ -- 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] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
On Sun, 14 Nov 2010, Gordon Henderson wrote: On Sun, 14 Nov 2010, Gordon Henderson wrote: On Sun, 14 Nov 2010, Tzafrir Cohen wrote: On Sun, Nov 14, 2010 at 04:38:25PM +, Gordon Henderson wrote: Well, just to follow this up - it looks like there is no DAHDI and BRI support in asterisk 1.4 at all. libpri has support in 1.6, but not 1.4, so it's mISDN for the time being. Right. But there are patches with a backport from 1.6.0 floating around. Interested? Possibly - it might make life easier than trying to work through the maze of mISDN integration with 1.4... However this is to go back into a production box that's currently running 1.2 + mISDN and I have a one-day window of testing opportunity... :) Think I've found the patch at: https://issues.asterisk.org/view.php?id=14871 it doesn't go cleanly into 1.4.35 which is what I'm using right now, (2 fails), but I think I can fix it by hand. Hardware detects fine with the wcb4xxp module. Well, my window of opportunity to get this going is rapidly diminishing, although I'll have tomorow morning as well - however I can't get it to work. Although the channels looks like they're up: dsx*CLI dahdi show status Description Alarms IRQbpviol CRC4 B4XXP (PCI) Card 0 Span 1OK 0 0 0 B4XXP (PCI) Card 0 Span 2OK 0 0 0 B4XXP (PCI) Card 0 Span 3OK 0 0 0 B4XXP (PCI) Card 0 Span 4RED0 0 0 (only 3 lines connected) I get lots of logs complaining about unable to find the D channel and dialling out yields CHANUNAVAIL calling in gives BTs digital dot telling me it's temporarily out of order, so giving up and going back to my old mISDN and asterisk 1.2 setup... Ah well, I'll come back to in in 2 years time I guess when I'm looking at 1.8... Gordon -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
I guess it will not work with PSTN lines since the control is transferred to
the Exchange. I am not too sure, I am just sharing my thoughts
On Fri, Nov 19, 2010 at 9:28 PM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi Gopalakrishnan A.N,
I tried it but it seems like my telco is overwriting the value I set as
callerid.
Maybe it is possible with Voip providers only.
Giorgio Incantalupo
Gopalakrishnan A.N wrote:
Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id checkbox while creating VoIP trunk then it
started working for me..
On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N sai...@gmail.com
mailto:sai...@gmail.com wrote:
Please try this in your dialplan
Set(CALLERID(name)=${CALLERID(num)})
Some where I tried and it worked with VoIP account A to B as VoIP
trunk and B forward the call to C whereas in C A's number will be
displayed.
If you could paste more details as Danny said that would help the
list to assist you more.
On Fri, Nov 19, 2010 at 9:11 PM, Danny Nicholas da...@debsinc.com
mailto:da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when
transferring call from
ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has
transferred the
call as if Charlie is receiving the call from Alice,
transparently.
Tried to set the callerid but Charlie sees my telco line
number, not the
callerid of Alice.
How can I do this?
Thank you.
Giorgio
--
We know that Alice and Charlie are both on external trunks.
We DON'T know
what flavor of Asterisk you are using, but it probably doesn't
matter your
call is going like this
ID #1 -- asterisk -- destination.
If destination were internal, ID#1 would remain intact, but
since you are
opening a new trunk to forward the call, you lose ID#2 and
replace it with
your Telco ID. You could spoof this depending on your asterisk
version/telco arrangement, but by default, things are as you
described.
--
_
-- 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
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com
mailto:sip%3asai...@gtalk2voip.com sip%253asai...@gtalk2voip.com
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.commailto:
sip%3asai...@gtalk2voip.com sip%253asai...@gtalk2voip.com
--
_
-- 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:
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--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com
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[asterisk-users] FFA (Fax For Asterisk) tif file (size) problem
Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error FAX handle 0: failed to queue document 'filename.tif', so we set it to 1680x2285, but it's still rejected. Is there a way to debug this further and fix it? We often have tif source files that we prefer to send, without converting to pdf and back to tif. Thank you in advance, Michael -- _ -- 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] Make call in AMI.
Good afternoon list. I need to make calls via AMI, but I need to leave the links in their respective contexts, to mobile phone calls by leaving out the context of mobile and so on. Already configured the settings that way, but I do not like the the Action Originate do it. I tried several ways, none successfully. What came closer to work the way I need is this: action: originate channel: Local/04191028...@intermovel context: returnCall extension: *10198 priority: 1 async: true interMovel is my context. But the answer on the Asterisk console was this: [Nov 19 16:49:56] NOTICE[23371]: chan_local.c:655 local_alloc: No such extension/context 04191028...@intermovel creating local channel [Nov 19 16:49:56] NOTICE[23371]: channel.c:3854 __ast_request_and_dial: Unable to request channel Local/04191028...@intermovel [Nov 19 16:49:56] ERROR[3843]: pbx.c:8396 device_state_cb: Received invalid event that had no device IE [Nov 19 16:49:56] ERROR[3843]: app_queue.c:862 device_state_cb: Received invalid event that had no device IE I need to do the links go out into different channels according to what is configured in the dialplan and dynamically. I can make a call by calling the channel normally, thus: action: originate channel: DAHDI/g1/04191028897 context: returnCall extension: *10198 priority: 1 async: true Does anyone have any idea how to do? Thank you in advance, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- 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] Installing Asterisk to it's own directory
I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is unfamiliar territory to me I'm not exactly sure how to accomplish this? Ideally, I'd like to just dump the newly compiled 1.8 and all it's dependencies into a standalone directory (say /testing/asterisk or something) and update my init script to point to the new binaries. I also run a Sangoma USB FXO card and DAHDI for a POTS line that I would like to test as well, should it work with the pre-compiled binaries that are already there? (DAHDI, etc) I've never tried this before, and before I potentially break something I'd like to know if it's possible and how to implement it? Thanks, Stephen -- _ -- 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] SPA941 WMI not lighting up when natted
Hi, I'm experiencing the same problem. We have 2 office locations and the Asterisk server is at one of them. At the other location, all SPA941 access the Asterisk server over an Internet link. All phones are set to nat=yes at the remote location. So my problem is that the MWI doesn't work at the remote location. The Sipsak messages are sent properly, but it's sent to the internet IP of the remote location so it will never reach the phones. A mailbox= is defined for each of my extensions. Can it be a configuration problem with the SPA941? - Mike aka...@gmail.com -- _ -- 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] Installing Asterisk to it's own directory
Hi Stephen, That's what people do when building precompiled packages for certain distros (along with a few more things). I use to do the following when building packages (with a few more options): ./configure --prefix=/usr --sysconfdir=/etc make make install DESTDIR=/my/destination/directory That would create the complete installation structure under '/my/destination/directory' Regards, Stephen Brown wrote: I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is unfamiliar territory to me I'm not exactly sure how to accomplish this? Ideally, I'd like to just dump the newly compiled 1.8 and all it's dependencies into a standalone directory (say /testing/asterisk or something) and update my init script to point to the new binaries. I also run a Sangoma USB FXO card and DAHDI for a POTS line that I would like to test as well, should it work with the pre-compiled binaries that are already there? (DAHDI, etc) I've never tried this before, and before I potentially break something I'd like to know if it's possible and how to implement it? Thanks, Stephen -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- 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] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Depending on what telco Charlie is connected to would change the CallerId presented to Charlie from being Alice's or Bob's Cid. When a call is forwarded, Charlie's telco receives different ANI and CID : some (seems to) favor ANI and some CID. An interesting thing to test is to let Bob issue a simple call to Charlie using a fake CID such as 0123456789. Will Charlie's phone display this non-existent number or 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
