Re: [asterisk-users] Patched Res_Musiconhold.So module

Appreciate the link. That still didn't resolve it. Any other suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Patched Res_Musiconhold.So module

Hi, For some reason, my Mac refuses to connect. It mentions that the server refused a client certificate offered. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] [CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday

Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC We have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and

Re: [asterisk-users] Patched Res_Musiconhold.So module

Hi, I created an account but when I go to issues.asterisk.org It still asks for a client certificate. See this screen shot, hopefully it showswhat I mean. http://firestar-hosting.com/clientcert .png--

Re: [asterisk-users] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk

Sorry. I wasn't sure if my first messaged was going to be approved, so posted twice by mistake. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk

Good day Asterisk users, If this is the wrong place to post this, my apologies. However, I'm trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after any announcement is played in the queue, or if

[asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC

I finally thought it might be a good time to start looking at the pjsip implementation in Asterisk 13. But trying to register to a sip cluster that uses SRV records fails randomly with: [Nov 4 15:50:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to send Request msg REGISTER/cseq=17800

[asterisk-users] [CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded

Contact === For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock <dan...@pocock.pro> Ralph Meijer <ral...@ik.nu>

[asterisk-users] Asterisk in the RTC Quick Start Guide

and RTC environment and whether this book makes it easier for people. Regards, Daniel 1. http://rtcquickstart.org 2. http://danielpocock.com/rtc-quick-start-becoming-a-book-now-in-beta -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Update peer IP address

rt opened. Daniel > Am 14.09.2015 um 21:51 schrieb Marie Fischer <ma...@vtl.ee>: > > > On 14.09.2015, at 21:58, Sebastian Kemper <sebastian...@gmx.net> wrote: > >> So I got rid of the firewall rule that opened the RTP ports. And then it >> dawned o

Re: [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

Hello Sam, Do you have any recommendation to overcome these NAT issues? On 8/14/15, Sam Basan sba...@bluebe.net wrote: Hi, It's looks like you are having NAT problem. Packets from the provider fail reaching your box. נשלח מטלפון נייד בתאריך 14 באוג' 2015 15:56,‏ Daniel - Asterisk earohua

[asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call.

Re: [asterisk-users] How to configure through the GUI 35 cisco ip phones, -spa502g (jg)

Thanks I will take a look. On 31/07/15 00:00, asterisk-users-requ...@lists.digium.com wrote: Re: How to configure through the GUI 35 cisco ip phones -spa502g (jg) -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] How to configure through the GUI 35 cisco ip phones -spa502g

Hi, I am new to Asterisk and VOIP so I am trying to find a decent howto guide on setting up cisco ip spa502g VOIP phones. I have found a interesting document on Cisco website but unable to access it.

Re: [asterisk-users] Update peer IP address

PM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Scott, I have changed the configuration as said it and will test it. I’m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without

Re: [asterisk-users] Update peer IP address

(with insecure=port,invite). If the provider is requiring you to accept invites from random IP addresses, get a new provider. On Thu, Apr 2, 2015 at 3:23 PM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Okay, Scott, I think we are on the wrong path. Maybe I'm

Re: [asterisk-users] Update peer IP address

01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net: On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: On 4/1/15 10:48 AM, Daniel Heckl wrote: John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk

Re: [asterisk-users] Update peer IP address

=invite,port makes any difference either (without alllowguest on). ​ On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place

Re: [asterisk-users] Update peer IP address

be normal... Daniel Am 31.03.2015 um 22:45 schrieb Scott Griepentrog sgriepent...@digium.com: You have two options for dealing with an IP change during the registration period: 1) set the registration time to shorter period of time to minimize the downtime 2) detect that the IP address

Re: [asterisk-users] Update peer IP address

John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk, not of my asterisk server. Kind regards, Daniel Am 01.04.2015 um 16:40 schrieb Tech Support aster...@voipbusiness.us mailto:aster...@voipbusiness.us: If I

Re: [asterisk-users] Update peer IP address

were even not found. I think there must be another solution. If I change insecure to insecure=port,invite - could that be a solution? Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? Daniel Am

Re: [asterisk-users] Update peer IP address

Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer? Am 31.03.2015 um 12:36 schrieb Daniel Heckl daniel.he...@gmail.com: Hello Sebastian

[asterisk-users] Update peer IP address

://stun.t-online.de/ nat=yes srvlookup=yes allowguest=no trustrpid=no insecure=invite qualify=yes Thank you! Daniel-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] How to perform some tasks after the CDR has been closed?

after that event? It would be nice if the channel is still available, since I need access to other channel variables. An alternative would be to pass those variables via the CDR if the channel has been deleted. Thanks, Daniel Gonzalez

Re: [asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?

, targeting to use ARI instead of database for making the integration of the two applications. Cheers, Daniel On 29/01/15 16:52, Matthew Jordan wrote: On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk 62...@mail.ru wrote: Hi all Have recently watched Matt Jordan's session on Kamailio World 2014

[asterisk-users] Finish extension (avoid dialplan to silently continue in the next priority of another extension)

? Thanks, Daniel -- _ -- 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

[asterisk-users] Log to file in Asterisk: append with newline

am trying: same = n,Set(FILE(/tmp/mylog.txt,,,al)=my-log-message) But this does not append (instead, it just overwrites - surprisingly, since the documentation says that this should append!). How can I append *and* make sure that a newline is added after the log message? Thanks, Daniel Gonzalez

[asterisk-users] Passing literals with commas to subroutine

, Daniel -- _ -- 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

[asterisk-users] Show Log(NOTICE) messages on the console

and regards, Daniel Gonzalez -- _ -- 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

[asterisk-users] WebRTC meeting Norfolk, 15 October 2014

I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody

[asterisk-users] WebRTC / Rejecting secure audio stream errors

I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works

[asterisk-users] force_avp ignored?

I'm using v11.11 I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with RTP/SAVPF and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or

Re: [asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration

want to use rtpenging for srtp(webrtc)-rtp(classic sip) gatewaying? If yes, maybe you can partition the users (classic-sip and webrtc-sip), then use two asterisk instances with routing via kamailio. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http

Re: [asterisk-users] Internal timing under load is critical ?

much more common. -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] chan_motif / res_xmpp problems

On 21/07/14 15:12, Daniel Pocock wrote: On 21/07/14 14:33, Joshua Colp wrote: Daniel Pocock wrote: I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP

Re: [asterisk-users] chan_motif / res_xmpp problems

On 22/07/14 18:20, Joshua Colp wrote: Daniel Pocock wrote: snip FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1 Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x releases? Nope. Is there any way I can enable ICE debugging? Not within 11

[asterisk-users] chan_motif / res_xmpp problems

I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list

Re: [asterisk-users] chan_motif / res_xmpp problems

On 21/07/14 14:33, Joshua Colp wrote: Daniel Pocock wrote: I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers

[asterisk-users] Dial international number over dahdi trunk

, but a real, working mobile german number) How can I dial international numbers via DAHDI? (in case it matters, my SS7 trunk provider is Telefonica España) Thanks! Daniel Gonzalez -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] chan_motify / res_xmpp bind address?

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it

Re: [asterisk-users] Getting source ip adress of incoming INVITE

On Fri, Jul 04, 2014 at 10:04:45AM -0400, Richard Kenner wrote: I'm interested in finding out what the source ip is of an invite in the dialplan (Asterisk 11). ${CHANNEL(recvip)} Doh! That is an obviouls place to look. I'm wondering why I didn't think about this or couldn't find any hints.

[asterisk-users] Getting source ip adress of incoming INVITE

I'm interested in finding out what the source ip is of an invite in the dialplan (Asterisk 11). As far as I can see this information isn't accessible. The only solution I can think of is parsing either Record-Route or Via headers. This is for recognizing guests in the default context for sip. --

[asterisk-users] Asterisk crashes suddenly

Hello friends, I have been experienced suddenly stops for my Asterisk server, I do not why is it happening. Asterisk's debug messages only tell me I have lacked g729 codec for translation to one peer minutes before the crashes occur [2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c:

Re: [asterisk-users] Asterisk call forward for T1 incoming calls

, the op referred to T1 channels. TBCT works on PRI, not simple T1. --Don Additionally, most providers charge for TBCT provisioning, so you would want to make certain that it is the correct way to get the functionality you are after. -- Daniel Taylor VP OperationsVocal

Re: [asterisk-users] Asterisk 1.6

is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711

[asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8

Hi All, I was wondering if it is possible to pass sound files to a macro as an argument in Asterisk 1.8? Thanks! Regards, Daniel van den Berg SureTel South Africa 087-944-7873 -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

. My initial thoughts are to maybe ask the mobile operator to switch off the voice mail functionality on those mobile phones and rather give a busy or engaged tone, but I would rather want to do this in Asterisk. Any help or advise on this matter will be greatly appreciated. Thanks! Daniel van den

Re: [asterisk-users] callfiles.call

On Fri, Jan 31, 2014 at 6:16 PM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, Hi i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06 MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1

Re: [asterisk-users] Asterisk 12 questions

On Thu, Jan 30, 2014 at 12:48 PM, Glen Millard glenmill...@gmail.comwrote: Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip instead of pjsip. Hi Glenn, I am missing something. I assumed that chan_sip was going to be added by default. Apparently not. I saw it in the

[asterisk-users] DTLS setting impacts encryption setting

If I understand correctly, setting encryption=no means that Asterisk will make outgoing calls without encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or not If encryption=yes, then Asterisk not only uses encryption for the outgoing calls

Re: [asterisk-users] AMI eventmask question

On Thu, Jan 23, 2014 at 9:05 PM, Daniel Jenkins dan.jenkin...@gmail.comwrote: On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.comwrote: On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote: Thanks - I've been through that doc before and couldn't find

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

and compress, write compressed data). -- Daniel Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] AMI eventmask question

On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: Hi I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? Are you talking about the

Re: [asterisk-users] AMI eventmask question

what the docs say, and the docs are generated from the source code now, you can only specifiy those larger blocks, Hope I've helped, Dan -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Jenkins

Re: [asterisk-users] AMI eventmask question

On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.com wrote: On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote: Thanks - I've been through that doc before and couldn't find the info needed, which is why I went to the source code eventually. All events are

Re: [asterisk-users] Asterisk AMI - PHP or Node.js?

On Sat, Dec 28, 2013 at 11:32 AM, Shahid H shah...@gmail.com wrote: Hi, I would like to develop a Call Center Dialer (outbound and inbound calls) and it would use AMI method to communicate with Asterisk Server. A daemon would need to run in the background, would you recommend coding in PHP

Re: [asterisk-users] OT - Merry Christmas and a Happy and Prosperous 2014

Hi Josue, May God the Father of our Lord and Saviour Jesus Christ bless you and your loved ones always! On 12/24/2013 10:37 PM, Josué Conti wrote: Dear Ladies and Gentlemen how are you? I would like to wish everyone and all their families, may God continue to bless and always illuminating your

[asterisk-users] How to recognize the Telco provider on outgoing calls only by sounds?

Dear list: When I call an specific number on the PSTN, the provider who holds the destination number give back an specified sound just after admitting their incoming calls. Is there a way to allow Asterisk to compare sounds received to decide what is the Telco answering the call? I'm planning to

[asterisk-users] Asterisk 11.6.0 not starting up

Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with asterisk -vvc and service asterisk start. Starting process just stop and shows: Illegal instruction as final output. Looking at logs I fouind at

Re: [asterisk-users] Asterisk 11.6.0 not starting up

, on Compiler Flags menu you have to deselect BUILD NATIVE parameter. Then make, make install, make samples, make config Regards El 25/11/2013 11:49, Leandro Dardini escribió: On which kind of processor are you trying to run asterisk? Is it a real or emulated CPU? Leandro 2013/11/25 Daniel

Re: [asterisk-users] Asterisk 11.6.0 not starting up

2013/11/25 Daniel - Asterisk earohua...@gmail.com Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with asterisk -vvc and service asterisk start. Starting process just stop and shows: Illegal instruction as final

[asterisk-users] Adding SIP method MESSAGE to Allow header

= messages auth_message_requests = yes subscribecontext = users All my extensions are in the 'users' context, and my text message handling is in a context called 'messages'. Anyone see what I might be missing ? Thanks in advance for any ideas. Kind regards, Daniel

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

Hi there, Sounds like codec ptime mismatch...what codec are you using? If you are using g729 make sure that you and your provider is giving the same ptime. On 10/29/2013 11:55 AM, Stelios Koroneos wrote: On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: All, The users in our

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

Hi there, In other words you are maybe on 60ms and they are on 20ms or vice versa. Do a wireshark trace and see if the codecs and ptime agree on both sides otherwise you will get grabbled sounds. On 10/29/2013 02:49 PM, Daniel van den Berg wrote: Hi there, Sounds like codec ptime mismatch

Re: [asterisk-users] Which events is generated as Asterisk Manager logoff

On 14 Oct 2013 18:18, Paul Belanger paul.belan...@polybeacon.com wrote: On 13-10-14 12:31 PM, virendra bhati wrote: As I said, I am running a event capture program and it looks for Events and work on the basis of events. But some time it stop working so I want to auto-connect with asterisk

[asterisk-users] How to bind to ipv4 ipv6

Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist

Re: [asterisk-users] How to bind to ipv4 ipv6

, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both

Re: [asterisk-users] user list archive

Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my

Re: [asterisk-users] user list archive

Sorry :) On 09/27/2013 06:16 PM, Rusty Newton wrote: On Fri, Sep 27, 2013 at 11:14 AM, Daniel van den Berg aster...@suretel.co.za wrote: Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses

Re: [asterisk-users] How to bind to ipv4 ipv6

IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you

[asterisk-users] Asterisk - WHMCS Intergration

Cheers Daniel -- _ -- 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

Re: [asterisk-users] LUA

On 07/18/2013 08:56 AM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org

Re: [asterisk-users] Mailing a fax with mutt does not succeed

Hello everyone, I'd changed the server and mutt started working, but I'll test your advices and wil let you lnow ass soon as I can. Thank you! Elder On Mon, Jun 24, 2013 at 7:38 AM, Larry Moore lmo...@omninet.net.au wrote: On 22/06/2013 2:17 PM, Steve Edwards wrote: On Sat, 22 Jun 2013,

Re: [asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip

running on the same host as kamailio to bridge the rtp between external and internal networks. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

Re: [asterisk-users] Mailing a fax with mutt does not succeed

(or equivalent log file)? If so, mutt basically works and the messages should give some clues. (2) What happens if you call mutt without any attachments? I am using mutt in exactly the same way and it works. jg Am 19.06.2013 21:50, schrieb Daniel - Asterisk: Hi Andre: I added echo to provide

[asterisk-users] Mailing a fax with mutt does not succeed

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif Unsuccessful Asterisk Command:

Re: [asterisk-users] Mailing a fax with mutt does not succeed

...@gmail.comwrote: Probably Asterisk does not know where mutt is, specify it's path in your System command. On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com wrote: Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works

Re: [asterisk-users] Mailing a fax with mutt does not succeed

, ) in new stack Elder D. Arohuanca Lima - Peru On Wed, Jun 19, 2013 at 1:38 PM, Andre Courchesne voipfor...@gmail.comwrote: Why echo | ? Alsy are you sire of the content of ${FAXDEST} and ${tempfax}. Add some NoOp before. On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 04:24:21PM -0400, Andre Goree wrote: Thanks so much for your suggestions. I'm running 1.0.x (yes, archaic, and in fact my actual task is migrating this system to asterisk11+Freepbx -- very fun in and of itself without regards to this issue...but I digress), and so I

Re: [asterisk-users] Issue dialing out

Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial(Zap/1-1, zap/g1/1XX|20|tT) in new stack Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1 Jun 15 13:06:08 VERBOSE[30232]: -- Attempting native bridge of Zap/1-1 and

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On Sat, Jun 15, 2013 at 10:28:50AM -0600, Nunya Biznatch wrote: Answer - There's a couple reasons I'm thinking this way, which may be misguided so thanks for making me think about it. First is redundancy. Offloading the PRIs and analog phones from the primary PBX means if there's an issue, I

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote: Setting the CID did not work, unfortunately :( [...] I'm going to try another number that we have through them in hopes that it'll complete and I'll let you know if that works. Do you have any other suggestions on what you think

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On Fri, Jun 14, 2013 at 09:43:29AM -0600, Nunya Biznatch wrote: System will use G.722 for VoIP Phones. [...] 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. So why use g722? Just use your local g711 law and thus avoid the transcoding impact to/from the PSTN and calls

Re: [asterisk-users] Cut offs on outgoing SIP calls

Hey Philipp, I will try soon the new version and let you know. Currently my users are pointing to a PBX in my local-private network with no problems. When I use wireshark I see my internal peers trying to send the ACK packets 4 or 5 times until hangup, at the same time the PBX are requesting

Re: [asterisk-users] Extenxions Optimization

On Sun, Jun 09, 2013 at 10:30:45AM +0200, Olivier CALVANO wrote: We want optimize my extensions file conf on asterisk 11.4.0 : ; Destination: Gambia Type: Fixe exten = _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) [5 lines] ; Destination: Libya Type: Fixe exten =

[asterisk-users] Sample config files installed to /etc

The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer in sip.conf and enable TCP However, I'm not sure

[asterisk-users] md5secret, secret and ha1b hash calculation?

of every hash when a user is added/updated, but only ha1 is consulted by the authentication code. The ha1b is simply stored to avoid the hassle of resetting all passwords if support for ha1b is completed in future. Regards, Daniel

[asterisk-users] realtime sip.conf and templates

Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure

Re: [asterisk-users] realtime sip.conf and templates

On 06/06/13 15:51, Daniel Pocock wrote: Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display

Re: [asterisk-users] RHEL6 packages - SRTP support? [patch]

On 03/06/13 23:04, Daniel Pocock wrote: On 03/06/13 19:18, Jason Parker wrote: On 06/03/2013 12:03 PM, Daniel Pocock wrote: I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However

[asterisk-users] offline builds - mp3 [patch]

As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will

[asterisk-users] Google/XMPP and Asterisk/XMPP

Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to

[asterisk-users] blog about WebRTC + TLS + Asterisk 11

identical working systems. To get the demo running for the WebSocket client, I really only needed to change about 5 lines in sip.conf - all other configuration is the default - the more painful step is rebuilding the packages with SRTP support. Regards, Daniel

Re: [asterisk-users] offline builds - mp3 [patch]

On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though

Re: [asterisk-users] offline builds - mp3 [patch]

On 04/06/13 19:13, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote: On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands

Re: [asterisk-users] asterisk debian package and digium repository

On 07/08/12 23:11, Rusty Newton wrote: On 8/7/2012 7:27 AM, Paul Belanger wrote: On 12-08-07 03:31 AM, ml asterisk wrote: Hi, I used to install asterisk on debian squeeze with digium repository. The last build of asterisk available is 1.8.11.1. Is this repository discontinued ? Since

[asterisk-users] RHEL6 packages - SRTP support?

I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module

[asterisk-users] missing build dependency / mISDNutils-devel and other errors

Building from the source RPM I get an error mISDNuser-devel is needed I was able to obtain all the other build dependencies from EPEL 6, but that one doesn't appear to existing in EPEL or in packages.asterisk.org I then tried adding --nodeps to the rpmbuild command: rpmbuild

Re: [asterisk-users] RHEL6 packages - SRTP support?

On 03/06/13 18:46, Jason Parker wrote: The packages currently do not support SRTP. I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons (see

Re: [asterisk-users] RHEL6 packages - SRTP support? [patch]

On 03/06/13 19:18, Jason Parker wrote: On 06/03/2013 12:03 PM, Daniel Pocock wrote: I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons

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