Re: [asterisk-users] Mp3 for IVR prompts
On Sat, Oct 10, 2009 at 11:47 PM, Steve Edwards wrote: > On Sat, 10 Oct 2009, RSCL Mumbai wrote: > > > How should I convert my .wav prompts into aLaw, uLaw, G729 ? > > The standard Asterisk prompts are already available in a wide variety of > encodings. > > Try "googling" for "asterisk convert mp3 to wav" > > Some will suggest to use Asterisk. Besides appearing to use a sledgehammer > for a fly-swatter, I don't like using a mission critical resource when > there are better alternatives like sox -- especially for scripting. > > g729 will be a bit of a bitch, however. > > I cobbled up a script to use mpg123 (to convert from MP3 to WAV), sox, > and normalize. > > My IVR prompts are in .WAV format. Refering to your previous post : ` You should strive to have prompts available in all the channel encodings actually used by your system. I have systems that only use ULAW, so all of my prompts are encoded as ULAW. (Sometimes I "cheat" and use WAV files since they are easier to work with and transcoding from WAV to ULAW is "cheap.") ` Can I convert my .WAV IVR greetings, MOH and other recordings into G729 format to prevent transcoding and hence CPU usage ? Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method to use SOX inside a Dialplan
> Steve Edwards wrote:
>>
>> The dialplan snippet would look something like:
>>
>> exten = s,n,system(/path-to-sox/sox a.wav b.wav c.wav)
>>
>> This would copy a.wav followed by b.wav to a new file, c.wav.
On Sat, 10 Oct 2009, Barton Fisher wrote:
> hmm, no luck. Here's what I have:
>
> exten => append,1,Noop(${PHRASEID}) ; this is the full path to original
> message without extension .wav
I'm on a "personal mission" to stamp out the abuse of the noop()
application since there is a more versatile application, verbose(),
specifically for this task :)
> exten => append,n,System(sox ${PHRASEID}.wav ${TEMPMESSAGE}.wav
> ${PHRASEID}.wav)
> -- Executing [app...@record:5] System("SIP/8001-0a0227c8", "sox
> /var/lib/asterisk/sounds/custom/7146762004/1_17692.wav
> /var/lib/asterisk/sounds/custom/7146762004/temp.wav
> /var/lib/asterisk/sounds/custom/7146762004/1_17692.wav") in new stack
sox\
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav\
/var/lib/asterisk/sounds/custom/7146762004/temp.wav\
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav
The output file has the same name as one of the input files. If I execute
"sox a.wav b.wav a.wav" I get "sox WARN wav: Premature EOF on .wav input
file." Curiously, if I execute "sox a.wav b.wav b.wav" it works, but that
is not what you want. In general, specifying an output file name the same
as an input file name is a bad idea because the result is inconsistent
from program to program.
What happens if you execute your command from a shell command line?
Is the path to sox in the environment of the process created by Asterisk?
If you append ">/tmp/stdout 2>/tmp/stderr" to your system() command line
you may get more clues.
--
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] outgoing sip calls work; incoming calls fail
On Sun, 11 Oct 2009 02:11:47 +0200 Ivan Stepaniuk wrote: > listm...@websage.ca wrote: > > > > On the LAN side I can see the INVITE and OKAY messages which end > > with a CANCEL, apparently initiated by the Asterisk gateway. > > > > On the WAN side I can see that my Asterisk gateway is repeatedly > > sending OKAY messages in response to the INVITE from my ITSP. I > > assume the trouble is that these messages are either not getting > > back to my provider or something is blocking the confirmation from > > them. This more or less confirms what was seen in the sip debug > > trace as well. > Post that SIP message from the CLI (sip debug), try adding > "externip=XXX.XXX.XXX.XXX" (your external/public IP address) to your > sip.conf global section, asterisk may be including it's private > address in the OKAY sent to your provider. > > Here's the last message in sip debug before it gives up: ... Retransmitting #6 (no NAT) to 66.51.127.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte Record-Route: From: "2508864577" ;tag=9Z5N4eayXp3Qm To: ;tag=as32af6364 Call-ID: b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE User-Agent: Asterisk PBX 1.6.0.15 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 992672626 992672626 IN IP4 96.50.76.138 s=Asterisk PBX 1.6.0.15 c=IN IP4 96.50.76.138 t=0 0 m=audio 15550 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '4e8ef1b977bf0e062212334634080...@192.168.11.1' in 6400 ms (Method: INVITE) ... 66.51.127.173 is my provider's SIP server 66.51.127.163 is my provider's RTP server I even check DNS to make sure both forward and reverse records jive. Externip was a good suggestion, and worth a try, though because I'm registering with my provider and using dynamic=yes, wouldn't they just reply to that anyway, especially given that the registration works fine? Anyway, after adding externip= to [general] and doing a sip reload in the console the problem remains... GM -- Greg Maruszeczka ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
On 10/11/09 01:27, Ivan Stepaniuk wrote: >Joseph wrote: >> I've tried setting my asterisk dtmf to rfc2833, inband it is not working. >> The other Asterisk Linksys is set dtmf = auto >If understand correctly, you have two asterisk servers and when you dial >from one the other, DTMF is not recognized. I also asume you are using >SIP to connect them as you mentioned dtmfmode. In any case, this should >be set to the same value on both sides, both rfc2833, or both info. You >don't wand inband and auto is just rfc2833 with automatic inband fallback. > >-- >Iv?n Stepaniuk >Alba Fot?nica S. L. >www.albafotonica.com Thank for the feedback. You are correct, I have two asterisk servers and dialing from one to another, using Linksys (supura unit) and sip. I'll double check the dtmf on them tomorrow. As on my Asterisk (have two lines) and dialing from one line to another (it is like dialing from one asterisk to another) I dtmf tone is recognized so I can access voicemail. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
Ivan Stepaniuk wrote: > listm...@websage.ca wrote: > >> On the LAN side I can see the INVITE and OKAY messages which end with a >> CANCEL, apparently initiated by the Asterisk gateway. >> >> On the WAN side I can see that my Asterisk gateway is repeatedly >> sending OKAY messages in response to the INVITE from my ITSP. I assume >> the trouble is that these messages are either not getting back to my >> provider or something is blocking the confirmation from them. This more >> or less confirms what was seen in the sip debug trace as well. >> > Post that SIP message from the CLI (sip debug), try adding > "externip=XXX.XXX.XXX.XXX" (your external/public IP address) to your > sip.conf global section, asterisk may be including it's private address > in the OKAY sent to your provider. > Sorry, "general" section. not global. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: > > On the LAN side I can see the INVITE and OKAY messages which end with a > CANCEL, apparently initiated by the Asterisk gateway. > > On the WAN side I can see that my Asterisk gateway is repeatedly > sending OKAY messages in response to the INVITE from my ITSP. I assume > the trouble is that these messages are either not getting back to my > provider or something is blocking the confirmation from them. This more > or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding "externip=XXX.XXX.XXX.XXX" (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
On Sun, 11 Oct 2009 01:41:36 +0200 Ivan Stepaniuk wrote: > listm...@websage.ca wrote: > > After running for months without issue I've got a situation where > > incoming SIP calls to my asterisk server are failing while outbound > > calls appear to be working as expected. > > > > The server is a gateway between my home LAN and a broadband cable > > connection with a dynamic IP. The gateway runs FreeBSD 7.1 and > > Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no > > problem. Outgoing calls can be made successfully and no error > > messages or warnings are reported by Asterisk. > > > > However, incoming calls appear to enter my dialplan as desired and > > go so far as to start ringing my SIP phone (Grandstream GXP-2000) > > but drop after two rings. The caller gets a busy tone and that's > > it. If I answer the call before the two rings I just get a moment > > of dead air and it drops in the same way. > > > > In the asterisk console (and log file) I see these messages at the > > fail point: > > > > [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: > > Maximum retries exceeded on transmission > > b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical > > Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] > > WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call > > b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical > > packet (see doc/sip-retransmit.txt) > > > > Okay, so I verified that my firewall is properly accepting traffic > > on the range of SIP and RTP ports as specified by my ITSP. > > > > After sending them a sip debug trace my provider said this: > > > > "It appears that your machine is not receiving replies when it > > tries to acknowledge the incoming call back to our server. This > > could be a firewall issue or potentially something else that > > changed without your knowledge." > > > > Furthermore, they suggested I might try registering and connecting > > directly to their Asterisk using only the Grandstream phone. I tried > > this and...surprise! Both inbound and outbound calls work fine but > > leave me without voicemail or any other services my PBX would be > > providing. > > > > Right, so now I'm thinking there must be something wrong with my > > Asterisk configuration yet I've made no config changes that would > > account for the sudden (and consistent) incoming call failures. > > > > Here's the relevant portions of my sip.conf if it helps (with > > credentials and ips replaced by Xs): > > > > [general] > > alwaysauthreject=yes > > dtmfmode=auto > > disallow=all > > allow=ulaw > > > > register => :x...@xxx.xxx.xxx.xxx:5060 > > register => :x...@xxx.xxx.xxx.xxx:5060 > > > > [101] > > type=friend > > context=websage > > host=dynamic > > deny=0.0.0.0/0 > > permit=XXX.XXX.XXX.XXX/24 > > qualify=yes > > secret= > > mailbox=...@default > > accountcode=101 > > > > Does your asterisk server have two network interfaces, one with a > private IP address and another one with the public one? > Did you try adding "canreinvite=no" to your 101 friend sip entry? > What does the SIP debug say? > Thanks for the response! Yes, the server has two interfaces with private addressing on the LAN side and a dynamically assigned IP on the public side (which has remained the same throughout this period). I did try canreinvite=no but it made no apparent difference in behaviour. In addition to the sip debug I've also performed tcpdump captures (using `tcpdump -i -n -s0 -v udp port 5060`) on both LAN and WAN sides: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. I've opened up all outbound udp and turned up logging on PF but see no evidence of any dropped or rejected traffic on my end (except the usual port scanning and netbios garbage on the outside). I'm stumped. GM -- Greg Maruszeczka Office: 250.412.9568 || Mobile: 250.886.4577 Skype: websage.ca || GTalk IM: gmarus http://websage.ca GnuPG-ID: 0x4309323E, http://pgp.mit.edu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method to use SOX inside a Dialplan
Steve Edwards wrote:
On Sat, 10 Oct 2009, Bart Fisher wrote:
I'm trying create a feature that allows a callers to add more speech to
his recording. I think this can be done inside a dialplan, but I can't
find an example of how to do this.
Basically,after he records the primary message, a menu would play asking
if he wants to append to this message. If yes, then he would record a
temp file with the additional message and when done, I want SOX to add
the temp message to the primary message making it one larger message.
Would you mind showing me an example of how to run SOX inside the
dialplan?
The system() dialplan application will allow you to run any executable.
If you plan on concatenating more than 32 input files you'll have to make
sure you have sox v14.3.0 or later.
The dialplan snippet would look something like:
exten = s,n,system(/path-to-sox/sox a.wav b.wav c.wav)
This would copy a.wav followed by b.wav to a new file, c.wav.
I would code the entire feature up as an AGI so you can hide all the ugly
details like creating files with unique file names, maybe running
normalize on the pieces before concatenation, error handling, maybe even
trimming the leading and trailing silence off each file so the gaps are
consistent, allowing the caller to listen to the new file and accept or
re-record the "suffix," cleaning up in case the caller hangs up, etc, etc,
etc.
hmm, no luck. Here's what I have:
exten => append,1,Noop(${PHRASEID}) ; this is the full path to original
message without extension .wav
exten => append,n,Playback(custom/dax/record)
exten =>
append,n,Set(TEMPMESSAGE=/var/lib/asterisk/sounds/custom/${IVR-EXTEN}/temp);this
is the full path to temp message without extension .wav
exten => append,n,Record(${TEMPMESSAGE}:wav|4|369) ; Begin recording new
message
exten => append,n,System(sox ${PHRASEID}.wav ${TEMPMESSAGE}.wav
${PHRASEID}.wav)
exten => append,n,system(rm ${TEMPMESSAGE}.wav)
exten => append,n,Goto(record,${IVR-EXTEN},confirm); go back and play
entire message
-- Goto (record,append,1)
-- Executing [app...@record:1] NoOp("SIP/8001-0a0227c8",
"/var/lib/asterisk/sounds/custom/7146762004/1_17692") in new stack
-- Executing [app...@record:2] Playback("SIP/8001-0a0227c8",
"custom/dax/record") in new stack
-- Playing 'custom/dax/record' (language 'en')
-- Executing [app...@record:3] Set("SIP/8001-0a0227c8",
"TEMPMESSAGE=/var/lib/asterisk/sounds/custom/7146762004/temp") in new stack
-- Executing [app...@record:4] Record("SIP/8001-0a0227c8",
"/var/lib/asterisk/sounds/custom/7146762004/temp:wav|4|369") in new stack
-- Playing 'beep' (language 'en')
-- Executing [app...@record:5] System("SIP/8001-0a0227c8", "sox
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav
/var/lib/asterisk/sounds/custom/7146762004/temp.wav
/var/lib/asterisk/sounds/custom/7146762004/1_17692.wav") in new stack
-- Executing [app...@record:6] System("SIP/8001-0a0227c8", "rm
/var/lib/asterisk/sounds/custom/7146762004/temp.wav") in new stack
-- Executing [app...@record:7] Goto("SIP/8001-0a0227c8",
"record|7146762004|confirm") in new stack
-- Goto (record,7146762004,39)
Anything wrong?
Bart
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Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: > After running for months without issue I've got a situation where > incoming SIP calls to my asterisk server are failing while outbound > calls appear to be working as expected. > > The server is a gateway between my home LAN and a broadband cable > connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk > 1.6.0.15 (built from ports) and registers to my ISTP no problem. > Outgoing calls can be made successfully and no error messages or > warnings are reported by Asterisk. > > However, incoming calls appear to enter my dialplan as desired and go so > far as to start ringing my SIP phone (Grandstream GXP-2000) but drop > after two rings. The caller gets a busy tone and that's it. If I answer > the call before the two rings I just get a moment of dead air and it > drops in the same way. > > In the asterisk console (and log file) I see these messages at the fail > point: > > [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum > retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a > for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. > [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging > up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical > packet (see doc/sip-retransmit.txt) > > Okay, so I verified that my firewall is properly accepting traffic on > the range of SIP and RTP ports as specified by my ITSP. > > After sending them a sip debug trace my provider said this: > > "It appears that your machine is not receiving replies when it tries to > acknowledge the incoming call back to our server. This could be a > firewall issue or potentially something else that changed without your > knowledge." > > Furthermore, they suggested I might try registering and connecting > directly to their Asterisk using only the Grandstream phone. I tried > this and...surprise! Both inbound and outbound calls work fine but > leave me without voicemail or any other services my PBX would be > providing. > > Right, so now I'm thinking there must be something wrong with my > Asterisk configuration yet I've made no config changes that would > account for the sudden (and consistent) incoming call failures. > > Here's the relevant portions of my sip.conf if it helps (with > credentials and ips replaced by Xs): > > [general] > alwaysauthreject=yes > dtmfmode=auto > disallow=all > allow=ulaw > > register => :x...@xxx.xxx.xxx.xxx:5060 > register => :x...@xxx.xxx.xxx.xxx:5060 > > [101] > type=friend > context=websage > host=dynamic > deny=0.0.0.0/0 > permit=XXX.XXX.XXX.XXX/24 > qualify=yes > secret= > mailbox=...@default > accountcode=101 > Does your asterisk server have two network interfaces, one with a private IP address and another one with the public one? Did you try adding "canreinvite=no" to your 101 friend sip entry? What does the SIP debug say? -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
Joseph wrote: > I've tried setting my asterisk dtmf to rfc2833, inband it is not working. > The other Asterisk Linksys is set dtmf = auto If understand correctly, you have two asterisk servers and when you dial from one the other, DTMF is not recognized. I also asume you are using SIP to connect them as you mentioned dtmfmode. In any case, this should be set to the same value on both sides, both rfc2833, or both info. You don't wand inband and auto is just rfc2833 with automatic inband fallback. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method to use SOX inside a Dialplan
On Sat, 10 Oct 2009, Bart Fisher wrote: > I'm trying create a feature that allows a callers to add more speech to > his recording. I think this can be done inside a dialplan, but I can't > find an example of how to do this. > > Basically,after he records the primary message, a menu would play asking > if he wants to append to this message. If yes, then he would record a > temp file with the additional message and when done, I want SOX to add > the temp message to the primary message making it one larger message. > > Would you mind showing me an example of how to run SOX inside the > dialplan? The system() dialplan application will allow you to run any executable. If you plan on concatenating more than 32 input files you'll have to make sure you have sox v14.3.0 or later. The dialplan snippet would look something like: exten = s,n,system(/path-to-sox/sox a.wav b.wav c.wav) This would copy a.wav followed by b.wav to a new file, c.wav. I would code the entire feature up as an AGI so you can hide all the ugly details like creating files with unique file names, maybe running normalize on the pieces before concatenation, error handling, maybe even trimming the leading and trailing silence off each file so the gaps are consistent, allowing the caller to listen to the new file and accept or re-record the "suffix," cleaning up in case the caller hangs up, etc, etc, etc. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method to use SOX inside a Dialplan
The record app has an append feature, if I remember correctly. Bart Fisher wrote: > I'm trying create a feature that allows a callers to add more speech to his > recording. I think this can be done inside a dialplan, but I can't find an > example of how to do this. > > Basically,after he records the primary message, a menu would play asking if > he wants to append to this message. If yes, then he would record a temp file > with the additional message and when done, I want SOX to add the temp message > to the primary message making it one larger message. > > Would you mind showing me an example of how to run SOX inside the dialplan? > > Thanks, Bart > > Alternatives: > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
Hardware echo usually helps. You can aslo try using OSLEC. - Original Message - From: B.Masoud @ SH To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, October 09, 2009 23:50 Subject: Re: [asterisk-users] choppy sound Hi, I am using CentOS Asterisk 1.4 The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware echo cancelation Does hardware echo will help? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a "restart when convenient" each morning around 4:00 AM. -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
If you are using freepbx, you have to change it in the GUI in the trunk
parameters.
B.Masoud @ SH wrote:
> Sorry for keep asking, but I did extensions reload, and restarted asterisk,
> What should the message looks like? I still get the same:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> "DAHDI/r0/0559857826|300|") in new stack
> -- Called r0/0559857826
>
> Thanks for your help.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
> Sent: Saturday, October 10, 2009 9:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] delay to dial
>
> B.Masoud @ SH wrote:
> > I have done the changes
> > exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
> >
> > I am getting this:
> >
> > -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> > "DAHDI/r0/0559857826|300|") in new stack
> > -- Called r0/0559857826
> >
> > Is it now on work? Or I have to restart?
> >
> It is not working. Issue an 'extensions reload' command at the asterisk
> CLI and try again. If it still does not work, then you have edited the
> wrong Dial. You should have tried that before asking in the list again.
>
> --
> Iván Stepaniuk
> Alba Fotónica S.L.
> www.albafotonica.com
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ___
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>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
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--
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How do
you spend it?
John Covici
cov...@ccs.covici.com
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Re: [asterisk-users] Method to use SOX inside a Dialplan
exten => _X.,n,System(sox arg1 ... argN) Martin On Sat, Oct 10, 2009 at 5:25 PM, Bart Fisher wrote: > I'm trying create a feature that allows a callers to add more speech to his > recording. I think this can be done inside a dialplan, but I can't find an > example of how to do this. > > Basically,after he records the primary message, a menu would play asking if > he wants to append to this message. If yes, then he would record a temp > file with the additional message and when done, I want SOX to add the temp > message to the primary message making it one larger message. > > Would you mind showing me an example of how to run SOX inside the dialplan? > > Thanks, Bart > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Method to use SOX inside a Dialplan
I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to add the temp message to the primary message making it one larger message. Would you mind showing me an example of how to run SOX inside the dialplan? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to Asterisk access voicemail - not working
Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk). PSTN to Asterisk is working, but not between two asterisk :-( I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing sip calls work; incoming calls fail
Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing calls can be made successfully and no error messages or warnings are reported by Asterisk. However, incoming calls appear to enter my dialplan as desired and go so far as to start ringing my SIP phone (Grandstream GXP-2000) but drop after two rings. The caller gets a busy tone and that's it. If I answer the call before the two rings I just get a moment of dead air and it drops in the same way. In the asterisk console (and log file) I see these messages at the fail point: [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt) Okay, so I verified that my firewall is properly accepting traffic on the range of SIP and RTP ports as specified by my ITSP. After sending them a sip debug trace my provider said this: "It appears that your machine is not receiving replies when it tries to acknowledge the incoming call back to our server. This could be a firewall issue or potentially something else that changed without your knowledge." Furthermore, they suggested I might try registering and connecting directly to their Asterisk using only the Grandstream phone. I tried this and...surprise! Both inbound and outbound calls work fine but leave me without voicemail or any other services my PBX would be providing. Right, so now I'm thinking there must be something wrong with my Asterisk configuration yet I've made no config changes that would account for the sudden (and consistent) incoming call failures. Here's the relevant portions of my sip.conf if it helps (with credentials and ips replaced by Xs): [general] alwaysauthreject=yes dtmfmode=auto disallow=all allow=ulaw register => :x...@xxx.xxx.xxx.xxx:5060 register => :x...@xxx.xxx.xxx.xxx:5060 [101] type=friend context=websage host=dynamic deny=0.0.0.0/0 permit=XXX.XXX.XXX.XXX/24 qualify=yes secret= mailbox=...@default accountcode=101 I'm now at a complete loss for how to proceed trying to resolve this and hoping someone with more experience than I on the list might have some ideas or suggestions. Any and all advice is warmly appreciated. Cheers, GM -- Greg Maruszeczka ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
> Sorry for keep asking, but I did extensions reload, and restarted asterisk,
> What should the message looks like? I still get the same:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> "DAHDI/r0/0559857826|300|") in new stack
> -- Called r0/0559857826
>
You should edit the 'Dial' command inside the macro-dialout-trunk
context, the line probably starts with "exten => s,19,Dial(..." as
stated in the CLI snippet you've posted, (s...@macro-dialout-trunk:19).
--
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com
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Re: [asterisk-users] delay to dial
Sorry for keep asking, but I did extensions reload, and restarted asterisk,
What should the message looks like? I still get the same:
-- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
"DAHDI/r0/0559857826|300|") in new stack
-- Called r0/0559857826
Thanks for your help.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Saturday, October 10, 2009 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
> I have done the changes
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
> I am getting this:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> "DAHDI/r0/0559857826|300|") in new stack
> -- Called r0/0559857826
>
> Is it now on work? Or I have to restart?
>
It is not working. Issue an 'extensions reload' command at the asterisk
CLI and try again. If it still does not work, then you have edited the
wrong Dial. You should have tried that before asking in the list again.
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] Mp3 for IVR prompts
On Sat, 10 Oct 2009, RSCL Mumbai wrote: > How should I convert my .wav prompts into aLaw, uLaw, G729 ? The standard Asterisk prompts are already available in a wide variety of encodings. Try "googling" for "asterisk convert mp3 to wav" Some will suggest to use Asterisk. Besides appearing to use a sledgehammer for a fly-swatter, I don't like using a mission critical resource when there are better alternatives like sox -- especially for scripting. g729 will be a bit of a bitch, however. I cobbled up a script to use mpg123 (to convert from MP3 to WAV), sox, and normalize. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
> I have done the changes
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
> I am getting this:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> "DAHDI/r0/0559857826|300|") in new stack
> -- Called r0/0559857826
>
> Is it now on work? Or I have to restart?
>
It is not working. Issue an 'extensions reload' command at the asterisk
CLI and try again. If it still does not work, then you have edited the
wrong Dial. You should have tried that before asking in the list again.
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] No sound on voicemail from analog line
Landy Landy wrote: >> Do you mean that incoming calls on your PSTN line works as >> they should, >> but not when they reach the voicemail? or that incomming >> calls on PSTN >> are always mute? >> > > Incoming calls on PSTN line work as they should but, when someone leaves a > voicemail message the messege is mute. When I try to retrieve the messeges I > get the prompt that says how many messeages are there Post the relevant part of your extensions.conf, * version, CLI output when the caller leaves a message and when you retrieve the message. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming extension not working.
On Fri, 9 Oct 2009, Ken D'Ambrosio wrote: > Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I > get: > > Rejected connect attempt from 64.2.142.19, who was trying to reach > '6031234567@' > > This leads me to my first question -- why doesn't it show a context? (My > second is, what's wrong with the snippets, below?): > > iax.conf: > [vitelity] > context=vitelity > register => username:passw...@inbound6.vitelity.net Asterisk doesn't show a context because it couldn't match the connection to any user section. 1) The "register" line belongs in the "general" section. 2) The "client section" needs to be named "vitel-inbound." Vitelity requires this. If you enable IAX2 debugging, you will see that Vitelity passes vitel-inbound as the username. 3) You didn't specify a type. 4) You didn't specify a secret. I like to keep the register keyword close to the client section so I use the following syntax: ; vitelity.net inbound [general](+) register= example:exam...@inbound6.vitelity.net [vitel-inbound] context = from-vitelity.net secret = example type= user This is the bare minimum. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mp3 for IVR prompts
On Sat, Oct 10, 2009 at 7:59 PM, Steve Edwards wrote: > On Sat, 10 Oct 2009, gergis.rasmy wrote: > > > can i use MP3 files as an IVR prompts directly without converting to > > .gsm format? > > You don't want to do this. > > Asterisk will attempt to use prompts encoded with the same codec being > used for the channel. So, unless you have a channel that is using MP3, > Asterisk would have to transcode the prompt every time it is used. Why > would you want to "burn" CPU cycles for this useless activity? > > You should strive to have prompts available in all the channel encodings > actually used by your system. I have systems that only use ULAW, so all of > my prompts are encoded as ULAW. (Sometimes I "cheat" and use WAV files > since they are easier to work with and transcoding from WAV to ULAW is > "cheap.") > > How should I convert my .wav prompts into aLaw, uLaw, G729 ? Thx Vai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
> I have done the changes
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
>
You need to type reload at the Asterisk console
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
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Re: [asterisk-users] delay to dial
I have done the changes
exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
"DAHDI/r0/0559857826|300|") in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, October 10, 2009 6:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
Ivan Stepaniuk wrote:
> John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
> seems to contain the channel technology, the 'w' should be inserted
> after the slash.
>
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
I agree.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
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Re: [asterisk-users] Grandstream GXP 2010 : multiple accounts not working
On Sat, 10 Oct 2009, jonas kellens wrote: > On my Grandstream GXP 2010 I have the possibility for 6 channels and > thus 6 different accounts... > > Line 1 I define an account that registers directly to an online > Asterisk-server, somewhere in a datacentre. > Line 2 I define an account that registers to the local Asterisk-server > (NSLU2 unslung) > > When I activate both accounts, only the first account (to the > Asterisk-server on the internet) registers. > When I only activate the first account, then the first account registers > well to the public Asterisk-server on the internet. > When I only activate the second, then the second account registers well > to the local Asterisk-server (NSLU unslung). > > Is it normal that I can not use both accounts at the same time ?! One > local and one to a public server ?? The times I've seen this fail have been when the phone is behind a router with a broken SIP ALG. Make sure the ALG is turned off, if possible, and look into using a STUN server to let them phone know how the NAT firewall is working if the remote isn't behind a proxy. And if you are already using STUN, make sure the phone isn't trying to use STUN for the internal connection - it's programmable on a per-account basis in the GXP phones. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP 2010 : multiple accounts not working
On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers. When I only activate the first account, then the first account registers well to the public Asterisk-server on the internet. When I only activate the second, then the second account registers well to the local Asterisk-server (NSLU unslung). Is it normal that I can not use both accounts at the same time ?! One local and one to a public server ?? When only the first account is enabled on the Grandstream IP-telephone, then the local Asterisk-server CLI shows this (when SIP debugging) : --- [Oct 10 18:09:21] Really destroying SIP dialog '0d8ad63d6c1940a11ff1ee2372846...@192.168.1.77' Method: OPTIONS [Oct 10 18:09:31] Reliably Transmitting (no NAT) to 192.168.1.100:5064: OPTIONS sip:n...@192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" ;tag=as1cad824f To: Contact: Call-ID: 64b39841529a74ad44a0b44403287...@192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:32] Retransmitting #1 (no NAT) to 192.168.1.100:5064: OPTIONS sip:n...@192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" ;tag=as1cad824f To: Contact: Call-ID: 64b39841529a74ad44a0b44403287...@192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:33] Retransmitting #2 (no NAT) to 192.168.1.100:5064: OPTIONS sip:n...@192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" ;tag=as1cad824f To: Contact: Call-ID: 64b39841529a74ad44a0b44403287...@192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:34] Retransmitting #3 (no NAT) to 192.168.1.100:5064: OPTIONS sip:n...@192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" ;tag=as1cad824f To: Contact: Call-ID: 64b39841529a74ad44a0b44403287...@192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:35] Retransmitting #4 (no NAT) to 192.168.1.100:5064: OPTIONS sip:n...@192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" ;tag=as1cad824f To: Contact: Call-ID: 64b39841529a74ad44a0b44403287...@192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:35] Really destroying SIP dialog '64b39841529a74ad44a0b44403287...@192.168.1.77' Method: OPTIONS Why is there an 'option' send to the local Asterisk-server when the local account on the Grandstream is disabled ?! Thanks for showing me some insight in all this ! Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
Ivan Stepaniuk wrote:
> John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
> seems to contain the channel technology, the 'w' should be inserted
> after the slash.
>
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
I agree.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
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Re: [asterisk-users] No sound on voicemail from analog line
> Do you mean that incoming calls on your PSTN line works as > they should, > but not when they reach the voicemail? or that incomming > calls on PSTN > are always mute? Incoming calls on PSTN line work as they should but, when someone leaves a voicemail message the messege is mute. When I try to retrieve the messeges I get the prompt that says how many messeages are there. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
John Novack wrote:
> B.Masoud @ SH wrote:
>
>> I use elastix,
>> I have this for dialout:
>>
>> exten => s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
>>
>> where should I add the w ??
>>
> right before the dialed number
> If I understand your code it should be:
>
> exten => s,8,Dial(www${OUT_${ARG1}}/${ARG2:${length}})
>
John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
seems to contain the channel technology, the 'w' should be inserted
after the slash.
exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
> I use elastix,
> I have this for dialout:
>
> exten => s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
>
> where should I add the w ??
>
>
right before the dialed number
If I understand your code it should be:
exten => s,8,Dial(www${OUT_${ARG1}}/${ARG2:${length}})
Remember that, from day one, Asterisk does NOT listen for dial tone from
the PSTN, and the fix isn't glamorous enough for some smart coder to fix
it, so the "w" has to be inserted to overcome this serious defect.
Subject of threads many times over the years, with no interest in fixing it.
Multiple w's can be inserted
> also what If I want 1 second delay?
>
2 w's as shown
Remember that this is absolute, and depending on the response time and
loading of your CO, it may or may not fix the problem.
More delay can be added, but that extends call setup time for every PSTN
call
John Novack
> thanks.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Saturday, October 10, 2009 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] delay to dial
>
> B.Masoud @ SH wrote:
>
>> Hello all,
>>
>> Is there anyway that I can configure Asterisk to start dialing out
>> from fxo after (xx) seconds from getting the dial tone? I don't want
>> tdm card to send the number immediately because it fails many times.
>>
>>
>
> You can use the w. This is from the wiki:
>
> If you need a .5 second pause while dialing a number you can insert a
> *w* in the appropriate place.
>
> Example:
>
> exten => _5XXX,n,Dial(ZAP/G1/w1269xxxw${EXTEN}${CALLERID(number)})
>
>
> This dials out G1, waits 1/2 second, dials the phone number and then
> waits 1/2 second again and then dial the extension along with the
> callerid number.
>
> Doug
>
>
>
>
>
>
>
> Checked by AVG - www.avg.com
> Version: 8.5.421 / Virus Database: 270.14.9/2426 - Release Date: 10/09/09
> 18:43:00
>
>
--
Dog is my co-pilot
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Re: [asterisk-users] delay to dial
I use elastix,
I have this for dialout:
exten => s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
also what If I want 1 second delay?
thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, October 10, 2009 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
>
> Hello all,
>
> Is there anyway that I can configure Asterisk to start dialing out
> from fxo after (xx) seconds from getting the dial tone? I don't want
> tdm card to send the number immediately because it fails many times.
>
You can use the w. This is from the wiki:
If you need a .5 second pause while dialing a number you can insert a
*w* in the appropriate place.
Example:
exten => _5XXX,n,Dial(ZAP/G1/w1269xxxw${EXTEN}${CALLERID(number)})
This dials out G1, waits 1/2 second, dials the phone number and then
waits 1/2 second again and then dial the extension along with the
callerid number.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
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[asterisk-users] paging/intercom
I'm having hard times with paging intercom
Heres my dialplan
exten => 777,1,Goto(intercom,777,1)
[intercom]
exten => 777,1,SIPAddHeader(Call-Info: \;answer-after=0)
exten => 777,2,Page(Local/3...@page& Local/3...@page& Local/3...@page)
[page] ; Paging context
exten => _X.,1,Macro(page,SIP/${EXTEN})
[macro-page]
; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
;
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten => s,2,SIPAddHeader(Alert-Info: Ring Answer) ; this one is for the
Polycom IP601
exten => s,3,Dial(${ARG1}|3|) ; should ring 3 seconds
exten => s,4,Hangup
exten => s,104,Hangup
Problem is that if lets say 310 is on the phone with a client.. and one
pages all.. (777) then the 310 phone (Linksys 942) puts current call on
hold and or drops the call to answer page.
Is that the send audio to speaker option in preference of the phone that's
not right ?
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Re: [asterisk-users] Mp3 for IVR prompts
On Sat, 10 Oct 2009, gergis.rasmy wrote: > can i use MP3 files as an IVR prompts directly without converting to > .gsm format? You don't want to do this. Asterisk will attempt to use prompts encoded with the same codec being used for the channel. So, unless you have a channel that is using MP3, Asterisk would have to transcode the prompt every time it is used. Why would you want to "burn" CPU cycles for this useless activity? You should strive to have prompts available in all the channel encodings actually used by your system. I have systems that only use ULAW, so all of my prompts are encoded as ULAW. (Sometimes I "cheat" and use WAV files since they are easier to work with and transcoding from WAV to ULAW is "cheap.") -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
>
> Hello all,
>
> Is there anyway that I can configure Asterisk to start dialing out
> from fxo after (xx) seconds from getting the dial tone? I don’t want
> tdm card to send the number immediately because it fails many times.
>
You can use the w. This is from the wiki:
If you need a .5 second pause while dialing a number you can insert a
*w* in the appropriate place.
Example:
exten => _5XXX,n,Dial(ZAP/G1/w1269xxxw${EXTEN}${CALLERID(number)})
This dials out G1, waits 1/2 second, dials the phone number and then
waits 1/2 second again and then dial the extension along with the
callerid number.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
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[asterisk-users] Slightly OT: Astricon and Google Wave
Looking at my shiny new Google Wave account, I was wondering if anyone else on this list is in the beta AND going to Astricon. Astricon seems like it would be a good test of the kind of collaboration GW is trying for. In any case, I'd love to try to do an Astricon wave so let me know if you're interested and we'll get together. I know at least two other people who'll be there presenting. This might be an interesting way for them to get feedback. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi GSM handover
The approach to Wi-Fi to GSM handover differs depending if its handled by the cellular carrier or enterprise PBX. If it's handled by the cellular carrier (as in the case of UMA) there's the advantage of not tying up additional trunk lines for incoming calls, but unless the rest of the enterprise is with the same carrier, there's no short-digit dialing to co-workers, mobile or at their desk. If it's handled by the PBX, then depending on what client is built on the smartphone, the end-user can access the key PBX features. Call control becomes a little trick. The general approach is that all calls that come into the enterprise PBX. If the end-user's phone is in Wi-Fi mode, a new VoIP-based call leg is established, either over the LAN, WAN, or Internet. If the end-user's phone is in cellular mode, the convergence appliance must create a new PSTN-based call leg. In either case, the software on the (smart)phone communicates either in-band or out-of-band (cellular data, usually) with the convergence appliance so that the appliance knows the connectivity state of the (smart)phone. When outbound calls are made, usually the (smart)phone client calls the PBX, and then patches it through. This provides a consistent call flow experience, including the ability for the organization to tracks calls (CDRs), the ability for the end-user to transfer calls, conference people in, etc. Both of the vendors I mentioned should have whitepapers/documentation on how their products work. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Sent: Saturday, October 10, 2009 3:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Wifi GSM handover Thank to Frank and Steve for your answers My understanding is that you need to place on operator premise an equipment that checks first the availability of the user on VoIP. If not registered, it's routing the call through the cellular network. Is it correct ? But during the handover (wifi to GSM), how does it works ? Is it the operator that initiates a call on the GSM network. If so, I guess the mobile device need to have some logic to seamlessly switch between the 2 channels, isn't it ? If it's the mobile device that initiates the call to the GSM network, it will also require some logic to do that. So my question is, is the handover something standard in every mobile device supporting GSM and VoIP or do you require an extra piece of software to do the trick ? Is this principal applies to every "transport technology", I mean VoIP through WIfi or VoIP over 3G ? Thanks in advance Patrick On Sat, Oct 10, 2009 at 06:57, Frank Bulk wrote: > There are two commercial vendors that come to mind, namely DiVitas and > Agito. > > Frank > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick > Sent: Friday, October 09, 2009 8:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Wifi GSM handover > > Hello guys, > > I'm wondering what is required and involved in order to provide a > wifi/GSM handover to customers. > After googling I haven't found any product/vendor. Do you have an idea ? > > Thanks in advance > Patrick > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] delay to dial
Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from getting the dial tone? I don't want tdm card to send the number immediately because it fails many times. Thanks for any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrond DTMF detection on Zap channel
I'm using Zap, not chan_local i've tried to record the call and have seen that the audio DTMF toned received is very poor, i've tried to put relaxdtmf=yes in zapata.conf and increare rxgain and txgain from 0 to 5 but it doesn't seems to be much better. Is there something else to do? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F wrote: > are you using chan_local? > try disabling the hardware DTMF. > > Sent using my wired Blueberry. > > On 10/9/09, nik600 wrote: >> Dear all >> >> i have a TE205P connected to an Asterisk 1.2.18. >> >> Yes i know, the version is old but since now the system was stable and >> i don't have the necessity of an upgrade. >> >> The system provide an IVR service that: >> >> 1) receive the call >> 2) verify the queue length >> 3) hangup if queue length is > 1 >> 4) put the call in the queue othervise >> >> Then, there is an AGI php script that >> 1) verify the queue >> 2) wait 5 seconds if the queue is empty >> 3) pick-up a call from the queue and transfer it to an extension othervise >> >> Finally, the extension lanuch another AGI php script that requires >> some DTMF tone to the user to perform some actions. >> This system is working properly since 2006. >> >> Well, the problem during last days is that it seems that sometimes the >> DTMF recognition doesn't work, in the debug i get: >> >> AGI Tx >> 200 result=0 >> >> But users complains to me because they assure to have digited >> something different than 0. >> The problem seems to be reproducible when the system is loaded (i >> don't have information on the SO but we receive abut 2500 calls per >> hour each call is very short because usually it is hangup after a very >> short time, as the queue length is very often 1) >> >> It's not an AGI application problem as i get the "wrong" dtmf tone >> directly from Asterisk. >> It's not a phone problem as the same phone may retry and then it works. >> >> Is it possible to relate it with the load of the server? >> >> Can you suggest me something? >> >> Thansk >> >> -- >> /*/ >> nik600 >> http://www.kumbe.it >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently?
Marco Mouta wrote: > Sad to say, but I believe this is only the small beginning…. Just a guess, and off-topic, but probably someone got very angry at citibank. At least in Spain, they (or a marketing contractor) seem to have called every single mobile phone in this country, they called me five times without even knowing I was already a customer. I bet they have the same marketing policies everywhere. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mp3 for IVR prompts
can i use MP3 files as an IVR prompts directly without converting to .gsm format? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing applications
You can try free version of MOR Softswitch with billing and routing: http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/ We rewrote Asterisk CDR completely and yes, it supports transfers. More info about MOR: http://www.voip-info.org/wiki/view/MOR Free version supports up to 10 simultaneous calls which is enough for majority of startups. You can check our manual to see what functionality is supported: http://wiki.kolmisoft.com/index.php/MOR_Manual Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: 2009 m. spalio 9 d. 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Billing applications Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi GSM handover
On Sat, Oct 10, 2009 at 10:16:43AM +0200, Patrick wrote: > Thank to Frank and Steve for your answers > My understanding is that you need to place on operator premise an > equipment that checks first the availability of the user on VoIP. If > not registered, it's routing the call through the cellular network. > Is it correct ? > But during the handover (wifi to GSM), how does it works ? Is it the > operator that initiates a call on the GSM network. If so, I guess the > mobile device need to have some logic to seamlessly switch between the > 2 channels, isn't it ? > If it's the mobile device that initiates the call to the GSM network, > it will also require some logic to do that. > So my question is, is the handover something standard in every mobile > device supporting GSM and VoIP or do you require an extra piece of > software to do the trick ? > Is this principal applies to every "transport technology", I mean VoIP > through WIfi or VoIP over 3G ? GSM calls are handled by an MSC (which is an SS7 switch) that talks to BSCs (basestation controllers) which talk to BTS (basestations), of course MSCs also talk to other MSCs. The GSM operator will have a UMA gateway in the network. A UMA phone will 'listen' for both GSM and WiFi and if it detects that the WiFi is 'known' it connects to that and it will connect through to the UMA gateway and the GSM network will switch the call to WiFi, if the user wanders off the WiFi area it will switch back to normal GSM operation. So the phone has to be UMA aware and the operator has to support it. On a normal GSM phone it is possible to write software that will switch calls between VoIP and GSM but you then generally have to control the endpoint of the call, so the GSM call usually goes through a VoIP access system and the software will switch the call to VoIP if it can, but the end-point is always the VoIP system that then calls the real number dialed. i.e. when the user dials a number it doesn't really go to that number directly, goes through the VoIP company who then can switch the transport in-between them and the handset. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi GSM handover
Thank to Frank and Steve for your answers My understanding is that you need to place on operator premise an equipment that checks first the availability of the user on VoIP. If not registered, it's routing the call through the cellular network. Is it correct ? But during the handover (wifi to GSM), how does it works ? Is it the operator that initiates a call on the GSM network. If so, I guess the mobile device need to have some logic to seamlessly switch between the 2 channels, isn't it ? If it's the mobile device that initiates the call to the GSM network, it will also require some logic to do that. So my question is, is the handover something standard in every mobile device supporting GSM and VoIP or do you require an extra piece of software to do the trick ? Is this principal applies to every "transport technology", I mean VoIP through WIfi or VoIP over 3G ? Thanks in advance Patrick On Sat, Oct 10, 2009 at 06:57, Frank Bulk wrote: > There are two commercial vendors that come to mind, namely DiVitas and > Agito. > > Frank > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick > Sent: Friday, October 09, 2009 8:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Wifi GSM handover > > Hello guys, > > I'm wondering what is required and involved in order to provide a > wifi/GSM handover to customers. > After googling I haven't found any product/vendor. Do you have an idea ? > > Thanks in advance > Patrick > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
