[asterisk-users] fromuser fromdomain
How can I force my users to be obliged to give a 'fromuser' and 'fromdomain' -parameter in their SIP-configuration ?? Is this set in the [general] -section of sip. conf ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to know AMI status
Dear All, I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me... Thanks, Velusamy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 Extensions.conf
Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local. kindly can any can help me to draw this dialpan in the extensions.conf Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 3RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:bs...@mg-tel.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Extensions.conf
Find my dahdi config files below dahdi-channels.conf ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 ClockSource group=0,11 context=default switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 ; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED group=0,12 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 32-46,48-62 context = default group = 63 ; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED group=0,13 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 63-77,79-93 context = default group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 group=0,14 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 94-108,110-124 context = default group = 63 Chan_dahdi.conf [trunkgroups] [channels] language=en context=default signalling = pri_cpe callwaiting=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no relaxdtmf=yes usedistinctiveringdetection=yes usecallingpres=yes busydetect=yes callprogress=yes rxgain=2.0 txgain=2.0 #include dahdi-channels.conf /etc/dahdi/system.conf # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 ClockSource span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED span=2,2,0,ccs,hdb3,crc4 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED span=3,3,0,ccs,hdb3,crc4 # termtype: te bchan=63-77,79-93 dchan=78 echocanceller=mg2,63-77,79-93 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 span=4,4,0,ccs,hdb3,crc4 # termtype: te bchan=94-108,110-124 dchan=109 echocanceller=mg2,94-108,110-124 # Global data loadzone= us defaultzone = us Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local. kindly can any can help me to draw this dialpan in the extensions.conf Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 3RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.
Re: [asterisk-users] E1 Extensions.conf
Hi, On Mon, Nov 09, 2009 at 12:52:15PM +0200, Khaled W Chehab wrote: Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi This is fairly simple. But I figure it is best that you actually understand what happens here. What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local. In this scenario we have several different Asterisk channels: 1. SIPcall - E1-port1 Incoming call from SIP generates a SIP channel. The dialplan context for it is set in sip.conf . You want it to generate a call to some DAHDI channel. This could be done using e.g. Dial(DAHDI/g1) (why g1? To what channels does it refer? See documentation in chan_dahdi.conf to see why I set it like that. Much of it is arbitrary). 2. E1-port1 - E1-port2 Loopback cable. It seems that you connected port 1 to port 4 rather than to port 2, right? 3. E1-port2 - sip extension Now we have an incoming DAHDI call. The dialplan context is set from 'context' in chan_dahdi.conf (where exactly?) . Now you'll probably need to use some dialplan such as: Dial(SIP/your-local-sip-extension) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know AMI status
velusamy velu wrote: Dear All, I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me... Did you make the necessary changes to manager.conf? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
Sendtext() works for SIP endpoints _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Saturday, November 07, 2009 9:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
That may not work for all sip phones. Some (like xlite/eyebeam) crash when receiving a text, others drop the subsequent call (Aastra 5x). These observations are based on a project we did in late 2008; so be sure to do a proof of concept before you get too deep into the project. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 09, 2009 9:12 AM To: Asterisk Users List Subject: Re: [asterisk-users] Text messaging Sendtext() works for SIP endpoints _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Saturday, November 07, 2009 9:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
What does Sendtext() actually do? Does it send a SIP request of method MESSAGE? What does it do on a hardware channel - say, analog or TDM? Michelle Dupuis wrote: That may not work for all sip phones. Some (like xlite/eyebeam) crash when receiving a text, others drop the subsequent call (Aastra 5x). These observations are based on a project we did in late 2008; so be sure to do a proof of concept before you get too deep into the project. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 09, 2009 9:12 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Text messaging Sendtext() works for SIP endpoints *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thomas Perron *Sent:* Saturday, November 07, 2009 9:39 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
It does nothing on hardware channels. SendText is just works on SIP channels. Purpose of SendText is showing text messages on user phone screen. show application SendText -= Info about application 'SendText' =- [Synopsis] Send a Text Message [Description] SendText(text[|options]): Sends text to current channel (callee). Result of transmission will be stored in the SENDTEXTSTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Text transmission not supported by channel At this moment, text is supposed to be 7 bit ASCII in most channels. The option string many contain the following character: 'j' -- jump to n+101 priority if the channel doesn't support text transport On Mon, Nov 9, 2009 at 4:50 PM, Alex Balashov abalas...@evaristesys.comwrote: What does Sendtext() actually do? Does it send a SIP request of method MESSAGE? What does it do on a hardware channel - say, analog or TDM? Michelle Dupuis wrote: That may not work for all sip phones. Some (like xlite/eyebeam) crash when receiving a text, others drop the subsequent call (Aastra 5x). These observations are based on a project we did in late 2008; so be sure to do a proof of concept before you get too deep into the project. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 09, 2009 9:12 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Text messaging Sendtext() works for SIP endpoints *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thomas Perron *Sent:* Saturday, November 07, 2009 9:39 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
I assumed the ATA/gateway would throw away or reject the message since I don't think there's an analog equivalent...but I'll wait for the analog experts to jump in. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, November 09, 2009 9:50 AM To: Asterisk Users List Subject: Re: [asterisk-users] Text messaging What does Sendtext() actually do? Does it send a SIP request of method MESSAGE? What does it do on a hardware channel - say, analog or TDM? Michelle Dupuis wrote: That may not work for all sip phones. Some (like xlite/eyebeam) crash when receiving a text, others drop the subsequent call (Aastra 5x). These observations are based on a project we did in late 2008; so be sure to do a proof of concept before you get too deep into the project. -- -- *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 09, 2009 9:12 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Text messaging Sendtext() works for SIP endpoints -- -- *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thomas Perron *Sent:* Saturday, November 07, 2009 9:39 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Extensions.conf
Hi,
I have a digium card (digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between spans on digium card in order to test
the spans.
I connect port 1 and port4 with cross E1 cable
I am trying to do this scenario
SIPcall-- Digium span 1---(Loop)Span 4sip
mailto:extens...@xx.xx.xx.xx extens...@xx.xx.xx.xx.
Kindly can you help me on how to forward the call from Span1-àSpan4 and then
from span4-à...@xx.xx.xx
My dahdi_channels.conf
; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER)
group=1
context=default
switchtype = euroisdn
signalling = pri_net
channel = 1-15,17-31
context = default
;group = 63
; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
group=4
;context=default
switchtype = euroisdn
signalling = pri_cpe
channel = 94-108,110-124
context = incomingck
;group = 63
-extensions.conf-
[default]
exten = _X.,1,Dial(DAHDI/G1/${EXTEN})
[incomingck]
exten = _X.,1,Dial(SIP/96123...@212.98.141.217,60)
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
___
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Re: [asterisk-users] Help with concurrent VoIP calls
Thanks for suggestions, everyone- I should have thought about jitter and latency as I began to use up more more bandwidth. I was concerned that it was a problem with my configuration of Asterisk, but it looks like is really is a bandwidth issue. By the way, Joe- I've been in another situation with my cableco Asterisk/VoIP (on a business connection!) and would frequently have trouble getting *one* call that sounded good, even though we had several megabits up down, with no other traffic on the network. Charter's service is horrible- there were several times pinging Google took over 1 second. John Timms On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote: Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging
Michelle Dupuis wrote:
I assumed the ATA/gateway would throw away or reject the message since I
don't think there's an analog equivalent...but I'll wait for the analog
experts to jump in.
It appears that Sendtext() simply invokes the callback stub
ast_channel_tech.send_text, and this is implemented by various channel
drivers in a proprietary manner.
In the case of chan_sip.c:sip_sendtext(), the implementation indeed
uses SIP MESSAGE:
static int transmit_message_with_text(struct sip_pvt *p, const char *text)
{
struct sip_request req;
reqprep(req, p, SIP_MESSAGE, 0, 1);
add_text(req, text);
return send_request(p, req, XMIT_RELIABLE, p-ocseq);
}
In the DAHDI implementation, it appears that some kind of acoustic
in-band tones are generated using main/tdd.c:tdd_generate(). I am not
certain what exactly the applicable standard is or how it works.
IAX2 appears to have a text frame type:
static int iax2_sendtext(struct ast_channel *c, const char *text)
{
return send_command_locked(PTR_TO_CALLNO(c-tech_pvt),
AST_FRAME_TEXT,
0, 0, (unsigned char *)text, strlen(text) + 1, -1);
}
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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[asterisk-users] local channels
I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context
where:
[my_context]
exten = my_priority,1,Answer()
exten = my_priority,n,Dial(${LOCAL_DIAL})
and LOCAL_DIAL has the actual phone number to dial.
The first call goes through just fine and I see DAHDI/1/ being
called. The second call I see
DAHDI/2/ and a message about everyone is busy on congested.
I presume I can have more than one local channel active? My AMI channel
line is:
Channel: Local/my_prior...@my_context for both calls. I have a Variable
with the LOCAL_DIAL set.
I am using DAHDI 2.2.0 with libpri 1.4.10.2 and asterisk 1.4.26.2
With a PRI connection.
All normal calls to phones work fine.
When I make my (2) local calls all 23 lines are idle.
Is there something I am missing? Why would I not be able to make 2 local
channel
calls at the same time?
Jerry
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Re: [asterisk-users] Text messaging
Hakan, I did not ask about the purpose of Sendtext() - I know the purpose, and on the level on which you have explained it, it is self-evident. I asked about how it was implemented underneath. Even in the context of SIP channels solely, there are numerous ways to send what one might term a message. -- Alex Hakan C wrote: It does nothing on hardware channels. SendText is just works on SIP channels. Purpose of SendText is showing text messages on user phone screen. show application SendText -= Info about application 'SendText' =- [Synopsis] Send a Text Message [Description] SendText(text[|options]): Sends text to current channel (callee). Result of transmission will be stored in the SENDTEXTSTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Text transmission not supported by channel At this moment, text is supposed to be 7 bit ASCII in most channels. The option string many contain the following character: 'j' -- jump to n+101 priority if the channel doesn't support text transport On Mon, Nov 9, 2009 at 4:50 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: What does Sendtext() actually do? Does it send a SIP request of method MESSAGE? What does it do on a hardware channel - say, analog or TDM? Michelle Dupuis wrote: That may not work for all sip phones. Some (like xlite/eyebeam) crash when receiving a text, others drop the subsequent call (Aastra 5x). These observations are based on a project we did in late 2008; so be sure to do a proof of concept before you get too deep into the project. *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 09, 2009 9:12 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Text messaging Sendtext() works for SIP endpoints *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thomas Perron *Sent:* Saturday, November 07, 2009 9:39 PM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* [asterisk-users] Text messaging IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] local channels
Jerry Geis wrote:
I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context
where:
[my_context]
exten = my_priority,1,Answer()
exten = my_priority,n,Dial(${LOCAL_DIAL})
and LOCAL_DIAL has the actual phone number to dial.
The first call goes through just fine and I see DAHDI/1/ being
called. The second call I see
DAHDI/2/ and a message about everyone is busy on congested.
I presume I can have more than one local channel active? My AMI channel
line is:
Channel: Local/my_prior...@my_context for both calls. I have a Variable
with the LOCAL_DIAL set.
That is correct.
It sounds like your need to make sure you're using the same trunk
group within DAHDI over and over:
Dial(DAHDI/1/${LOCAL_DIAL})
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] how to configure softphones in asterisk server
As I said, please keep discussion on list. aster...@opensourcesolution.in wrote: hi all, first of all i appologise for sending u pvt email. i have installed asterisk on Centos 5.3, plz open the attachment in which i had drawn a tolpology. i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.6 crashing
*Darrick Hartman:* NO! If you're using a specific 'branch' of asterisk, the latest release in that branch is the recommended version. There are almost certainly bugs/issues with earlier versions. 1.6.1.9 is the recommended version of Asterisk 1.6.1.x. *Danny Nicholas:* RC's are bleeding edge; x.x are considered stable, but you are almost always better off using the highest x.x stable release. Ok, I will try upgrading to 1.6.1.9 to see if that helps *Olivier* No, I didn't mean that : I only meant that I my particular case, that helped me to work around regular crashes (up to 5 times a day). It doesn't seem our problems are related then, I only have suffered 2 crashes in 1 month of use. Again, thanks for the help guys! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
Hi, He did that to me too (and previously). He's a complete fucking pain. I find it laughable that someone working for 'opensourcesolution' cant install a damned softphone. Clearly he is in the wrong business. Steve On 9 Nov 2009, at 16:32, Alex Balashov wrote: As I said, please keep discussion on list. aster...@opensourcesolution.in wrote: first of all i appologise for sending u pvt email. i have installed asterisk on Centos 5.3, plz open the attachment in which i had drawn a tolpology. i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
I would tend to concur. This is not an uncommon phenomenon on these lists and especially from that part of the world, however. People like this are not easily discouraged by criticism nor encumbered by any interest or sensivity in the prevalent ethics and culture of forums into which they plough. -- Sent from mobile device On Nov 9, 2009, at 12:03 PM, Steve Howes steve-li...@geekinter.net wrote: Hi, He did that to me too (and previously). He's a complete fucking pain. I find it laughable that someone working for 'opensourcesolution' cant install a damned softphone. Clearly he is in the wrong business. Steve On 9 Nov 2009, at 16:32, Alex Balashov wrote: As I said, please keep discussion on list. aster...@opensourcesolution.in wrote: first of all i appologise for sending u pvt email. i have installed asterisk on Centos 5.3, plz open the attachment in which i had drawn a tolpology. i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] got SIP response 482 Loop Detected back from xx.xxx.xxx.xxx
Hello everybody, This is my first post to this mailing list, so welcome everybody and thanks for the great community around asterisk. I few weeks back I got control over an asterisk server and was asked to create a number forwarding by the means of the configuration files. With the help of the asterisk book[1] and the people at #asterisk I changed and tested the extensions.conf and thought everyting was working. [1] http://www.asteriskdocs.org/ I made the following changes to create the forwarding: # diff extensions.conf for forwarding settings. http://debian.pastebin.com/da7a5d85 However sometimes *but* not always something goes horrible wrong and the connection drops. After further inquiring the other side tell me they see they are getting called but when they pick-up they hear a busy signal and the connection is dropped. The asterisk sip debug output on my side show a that there is some sort of loop detected followed by the destruction of the connection. My test call is from 0612182441 to 0208910330, and the wanted forwarding goes from 0208910330 to 0356929276. # an4705*CLI sip debug of failed call http://debian.pastebin.com/d5b98bbd4 # an4705*CLI sip debug of successful call http://debian.pastebin.com/d3027639c And this is my complete extensions.conf # cat /etc/asterisk/extensions.conf http://debian.pastebin.com/d348c9262 Can somebody help me? I am not an asterisk expert and issues like sometimes it works and sometimes it are harder to debug. What should I do to get a all time working forwarding. Thanks in advance, Best regards, Jelle asterisk-logs-and-settings-2009-11-09.tar.gz Description: application/gzip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to configure softphones in asterisk server
hi all, i have installed asterisk on Centos 5.3, plz i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Allow Header
Hi all, In the INVITE from my SIP provider to Asterisk i can see that the Allow Header includes an INFO Method, but Asterisk replies a 200 OK with an Allow Header without INFO Method. But in the RFC3261 (20.5) you can read: All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. My SIP provider seems to refuse to send SIP INFO DTMF and releases the call, because in 200 OK from * there is no INFO Method in the Allow Header. Is that correct. thx richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
That's what yahoo.answers.com is for! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, November 09, 2009 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to configure softphones in asterisk server You just don't get it, do you? Your indolent methods of getting what you want are not at your disposal here. This is not a homework help forum. -- Sent from mobile device On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote: hi all, i have installed asterisk on Centos 5.3, plz i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
You just don't get it, do you? Your indolent methods of getting what you want are not at your disposal here. This is not a homework help forum. -- Sent from mobile device On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote: hi all, i have installed asterisk on Centos 5.3, plz i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allow Header
Yes, it's correct. Asterisk needs to advertise its support of that method in order for the other UA to be willing to send messages with that request method to it. Coco Richard wrote: Hi all, In the INVITE from my SIP provider to Asterisk i can see that the Allow Header includes an INFO Method, but Asterisk replies a 200 OK with an Allow Header without INFO Method. But in the RFC3261 (20.5) you can read: All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. My SIP provider seems to refuse to send SIP INFO DTMF and releases the call, because in 200 OK from * there is no INFO Method in the Allow Header. Is that correct. thx richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreeBSD, ztdummy OHCI
I currently have an Asterisk running on an Alix 6B2 from PC Engines, but I am having trouble using ztdummy as a timing device. The USB driver is OHCI, and I believe ztdummy requires UHCI. So, I am wondering if there is a way to use a Kernel tick and ztdummy on FreeBSD, like it is possible on Linux? Play games at no cost with Tiscali Play - http://www.tiscali.co.uk/play ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile Voice setting
Hello all, I have successfully paired my mobile with asterisk, and chan_mobile already run very well, but sometimes when i restart asterisk chan_mobile fails to initialize with the error: chan_mobile.c: Incorrect voice setting for adapter toshiba, it must be 0x0060 - see 'man hciconfig' for details. I have tried several bluetooth adapters, as well as setting Class in /etc/bluetooth/main.conf to: 0x3e0100, 0x000100 and 0x006000 but the issue still happens, it usually pairs, but sometimes fail with the error mentioned. Thanks in advance, Ahmed Ossama ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
Jerry Geis wrote:
/ I am using the AMI to dispatch (2) calls to Local/my_priority at
my_context http://lists.digium.com/mailman/listinfo/asterisk-users
// where:
// [my_context]
// exten = my_priority,1,Answer()
// exten = my_priority,n,Dial(${LOCAL_DIAL})
//
// and LOCAL_DIAL has the actual phone number to dial.
//
// The first call goes through just fine and I see DAHDI/1/ being
// called. The second call I see
// DAHDI/2/ and a message about everyone is busy on congested.
//
// I presume I can have more than one local channel active? My AMI channel
// line is:
// Channel: Local/my_priority at my_context
http://lists.digium.com/mailman/listinfo/asterisk-users for both calls. I
have a Variable
// with the LOCAL_DIAL set.
/
That is correct.
It sounds like your need to make sure you're using the same trunk
group within DAHDI over and over:
Dial(DAHDI/1/${LOCAL_DIAL})
Alex,
My Dial() command is Dial($LOCAL_DIAL) and for the first call
it is DAHDI/1/ and for the second call it is DAHDI/2/XXX.
My LOCAL_DIAL has the complete dial command DAHDI/xx/
So I am using line 1 and line 2 of the PRI connection.
I dont see why the second call is saying - everyone busy or congested at
this time.
These are the only 2 calls active. One on line 1 and one on line 2.
Only 2 of the 23 lines available am I using. All 23 lines are in the
same group.
How do I tell why it thinks its busy? Thanks,
Jerry
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Re: [asterisk-users] local channels
I think the problem is that the way this works - if I'm not mistaken -
is that the attribute after the first delimeter in the channel string
is a trunk group and not a channel.
In other words, DAHDI/1 refers to circuit 1, not B-channel 1 of
circuit 1. B-channel 1 would be DAHDI/1/1.
Jerry Geis wrote:
Jerry Geis wrote:
/ I am using the AMI to dispatch (2) calls to Local/my_priority at
my_context http://lists.digium.com/mailman/listinfo/asterisk-users
// where:
// [my_context]
// exten = my_priority,1,Answer()
// exten = my_priority,n,Dial(${LOCAL_DIAL})
// // and LOCAL_DIAL has the actual phone number to dial.
// // The first call goes through just fine and I see DAHDI/1/
being // called. The second call I see
// DAHDI/2/ and a message about everyone is busy on congested.
// // I presume I can have more than one local channel active? My
AMI channel // line is:
// Channel: Local/my_priority at my_context
http://lists.digium.com/mailman/listinfo/asterisk-users for both
calls. I have a Variable // with the LOCAL_DIAL set.
/
That is correct.
It sounds like your need to make sure you're using the same trunk
group within DAHDI over and over:
Dial(DAHDI/1/${LOCAL_DIAL})
Alex,
My Dial() command is Dial($LOCAL_DIAL) and for the first call
it is DAHDI/1/ and for the second call it is DAHDI/2/XXX.
My LOCAL_DIAL has the complete dial command DAHDI/xx/
So I am using line 1 and line 2 of the PRI connection.
I dont see why the second call is saying - everyone busy or congested at
this time.
These are the only 2 calls active. One on line 1 and one on line 2.
Only 2 of the 23 lines available am I using. All 23 lines are in the
same group.
How do I tell why it thinks its busy? Thanks,
Jerry
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] local channels
My Dial() command is Dial($LOCAL_DIAL)
Perhaps you should be using:
Dial(${LOCAL_DIAL})
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Re: [asterisk-users] local channels
/ My Dial() command is Dial($LOCAL_DIAL)
/
Perhaps you should be using:
Dial(${LOCAL_DIAL})
Steve,
Thanks I tried that also and same result.
Jerry
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[asterisk-users] is an extension is use
Is there a way to tell if an extension is in use? We run a call center and it would be helpful for the people taking calls to see if we are on the phone or DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field but i will just turn on after a while even if the extension is not i use. the FOP in FreePBX doesn't appear to be that helpful. i am not sure what it is supposed to do. _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
This is what I see: -- Executing [my_prior...@my_context:1] Answer(Local/my_prior...@my_context-90d5,2, ) in new stack -- Executing [my_prior...@my_context:2] Dial(Local/my_prior...@my_context-90d5,2, DAHDI/3/4000) in new stack [Nov 9 16:25:17] WARNING[8979]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Local/my_prior...@my_context-90d5,2' status is 'CHANUNAVAIL' Normal calls all work just fine. I can call into the box and out the box to extensions and cell phones. When I place this call all lines are idle. My call through AMI is basically this: Action: Originate Async: yes Channel: Local/my_prior...@my_context Context: my_context Application: AGI Variable: LOCAL_DIAL=DAHDI/4/4001 Data: smvoice Priority: 1 Any ideas why the all channels busy and unable to create channel? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is an extension is use
You can use hints to tell If a line is inuse. There are built-in functions that do this also, but they don't always produce the desired result depending on what release you are on. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, November 09, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] is an extension is use Is there a way to tell if an extension is in use? We run a call center and it would be helpful for the people taking calls to see if we are on the phone or DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field but i will just turn on after a while even if the extension is not i use. the FOP in FreePBX doesn't appear to be that helpful. i am not sure what it is supposed to do. _ Hotmail: Trusted email with powerful SPAM protection. Sign up http://clk.atdmt.com/GBL/go/177141665/direct/01/ now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
LOCAL_DIAL is populated
- exten = s,1,Verbose(call ${LOCAL_DIAL})
- exten = s,2,Dial(${LOCAL_DIAL})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, November 09, 2009 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] local channels
/ My Dial() command is Dial($LOCAL_DIAL)
/
Perhaps you should be using:
Dial(${LOCAL_DIAL})
Steve,
Thanks I tried that also and same result.
Jerry
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Re: [asterisk-users] local channels
So 4001 is a local FXS DAHDI channel? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, November 09, 2009 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] local channels This is what I see: -- Executing [my_prior...@my_context:1] Answer(Local/my_prior...@my_context-90d5,2, ) in new stack -- Executing [my_prior...@my_context:2] Dial(Local/my_prior...@my_context-90d5,2, DAHDI/3/4000) in new stack [Nov 9 16:25:17] WARNING[8979]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Local/my_prior...@my_context-90d5,2' status is 'CHANUNAVAIL' Normal calls all work just fine. I can call into the box and out the box to extensions and cell phones. When I place this call all lines are idle. My call through AMI is basically this: Action: Originate Async: yes Channel: Local/my_prior...@my_context Context: my_context Application: AGI Variable: LOCAL_DIAL=DAHDI/4/4001 Data: smvoice Priority: 1 Any ideas why the all channels busy and unable to create channel? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allow Header
Hi Alex, i'm not sure to understand. Asterisk does support SIP INFO, so why doesn't Asterisk add the INFO Method in the 200OK Response? richard On Mon, Nov 9, 2009 at 6:38 PM, Alex Balashov abalas...@evaristesys.com wrote: Yes, it's correct. Asterisk needs to advertise its support of that method in order for the other UA to be willing to send messages with that request method to it. Coco Richard wrote: Hi all, In the INVITE from my SIP provider to Asterisk i can see that the Allow Header includes an INFO Method, but Asterisk replies a 200 OK with an Allow Header without INFO Method. But in the RFC3261 (20.5) you can read: All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. My SIP provider seems to refuse to send SIP INFO DTMF and releases the call, because in 200 OK from * there is no INFO Method in the Allow Header. Is that correct. thx richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call declined
Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten = 1234,1,Dial(SIP,gianca)* *exten = 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *- --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.* *--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=42ebb35e Content-Length: 0* * Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- ACK sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0* *- --- (7 headers 0 lines) --- dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=gianca,realm=asterisk,nonce=42ebb35e,uri= sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ,response=8d00b3e1b28ed2e40681a3a9ee410046,algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *- --- (13 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. Found user 'gianca' Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.116:5960 Found unknown media description format BV32 for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.116:5960 Looking for 12345 in tutorial (domain 192.168.1.100) list_route: hop: sip:gia...@192.168.1.116:14862* *--- Transmitting (no NAT) to 192.168.1.116:14862 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.116:14862
Re: [asterisk-users] Call declined
Try: [tutorial] exten = 1234,1,Dial(SIP/gianca,10,t) exten = 12345,1,Dial(SIP/giusy,10,t) You want a / between SIP and the name of the phone, not an ,. The 10 refers to the number of seconds you want the phone to ring. The t allows the channel to be transferred after pickup - not strictly needed, but I tend to put it in in most instances as generally you'll want it. For more information on the Dial application, see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo lombardo Sent: Tuesday, 10 November 2009 09:03 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call declined Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial [giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial extension.conf: [tutorial] exten = 1234,1,Dial(SIP,gianca) exten = 12345,1,Dial(SIP,giusy) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862http://sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265 v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. --- Reliably Transmitting (no NAT) to 192.168.1.116:14862http://192.168.1.116:14862 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=42ebb35e Content-Length: 0 Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- ACK sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100;tag=as29d2b71c From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0 - --- (7 headers 0 lines) --- dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862http://sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=gianca,realm=asterisk,nonce=42ebb35e,uri=sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100,response=8d00b3e1b28ed2e40681a3a9ee410046,algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265 v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 192.168.1.116 :
[asterisk-users] SendText
Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is an extension is use
Have you taken a look at the following? http://www.astassistant.com/ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, November 09, 2009 4:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] is an extension is use Is there a way to tell if an extension is in use? We run a call center and it would be helpful for the people taking calls to see if we are on the phone or DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field but i will just turn on after a while even if the extension is not i use. the FOP in FreePBX doesn't appear to be that helpful. i am not sure what it is supposed to do. Hotmail: Trusted email with powerful SPAM protection. Sign up now. http://clk.atdmt.com/GBL/go/177141665/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gradstream Budge Tone-201
Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
He wrote me too. I would have helped him, but the name on the email address threw me off. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 09, 2009 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] how to configure softphones in asterisk server That's what yahoo.answers.com is for! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, November 09, 2009 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to configure softphones in asterisk server You just don't get it, do you? Your indolent methods of getting what you want are not at your disposal here. This is not a homework help forum. -- Sent from mobile device On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote: hi all, i have installed asterisk on Centos 5.3, plz i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gradstream Budge Tone-201
On 10/11/09 1:12 PM, bilal ghayyad wrote: Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? I wouldn't recommend the BudgetTone - it's been a while since I used it, but there are better phones around (even from Grandstream). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is voicemail to text possible?
Hi, I understand that speech recognition technology is not very reliable, but skype has has launched a voicemail to text service, and googling showed that some other companies are also offering similar services. I haven't used any such service yet, but was curious is there any open source software available, which, to some extent, could help converting speech from voicemial wav files to text files and could be used with Asterisk? Or is there any other way to accomplish this? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is an extension is use
On 10/11/09 1:02 PM, Conklin, Tom wrote: Have you taken a look at the following? http://www.astassistant.com/ Also: http://www.asternic.org and the newer version: http://www.fop2.com -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
On 10/11/09 4:08 AM, C. Savinovich wrote: He wrote me too. I would have helped him, but the name on the email address threw me off. Poor guy - language/cultural barrier maybe? Here's some tips: 1. Read Asterisk The Future of Telephony (buy a copy or download from http://asteriskdocs.org) 2. Set up sip.conf/iax.conf based on what type of softphone 3. Download a softphone - I've listed a few here: http://www.venturevoip.com/news.php?rssid=2188 4. Make calls :D The most important step is number 1 - once you get the hang of Asterisk the rest will be easy :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension in use
There are a couple of ways you could see that, One would be by having a service .NET connected to the manager interface and watching for activity on the phone, this way you could tell if the phone is busy or not. [If phone has more than one line then set call-limit=1] Is this for routing purposes or just for display? The other thing you could use is Jabber. Look for OpenFire integration with asterisk and you'll see what I mean [google] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby wcse...@selbytech.com wrote: I think your featureLabel definition is wrong. On the login issue, ssh to the ip of the phone and login first with the user/pass you defined in the file (admin/123), then at the second login prompt use log/log. That should get you the log files which will show you your error. Thanks for the insight. After you mentioned that the syntax of the XML file may be wrong I looked around and found a more complete configuration I could find since mine was a copy and paste special. Using the new configuration the phone comes up but is unable register I *think* it may be an issue with NAT. When the phone fires up for the first time it tries to register for a while and the log didn't help much so I took a peak at the asterisk logging. It seems like packets are not getting back to the phone. I've enabled NAT in the configuration similar to how the other phones are configured but no dice. Note that the Asterisk device is not NATed but the phones are behind a NAT device. I get multiple of the following message in the phone: ERR 16:40:16.273722 JVM: %REG send failure: REGISTER On the asterisk server I keep getting NAT retries: Retransmitting #4 (NAT) to 71.226.175.137:1026: OPTIONS sip:1...@ip of NAT device:1027;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP ASTERISK IP:5060;branch=z9hG4bK53121c03;rport From: asterisk sip:aster...@209.251.157.91;tag=as5b0b32f5 To: sip:1...@ip of NAT:1027;user=phone;transport=udp Contact: sip:aster...@209.251.157.91 Call-ID: 090e1e583f29f9f000dd30ff5719f...@209.251.157.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Nov 2009 02:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Below is the full XML config for the phone: device xsi:type=axl:XIPPhone ctiid=9044468655 deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool dateTimeSetting dateTemplateM/D/Ya/dateTemplate timeZoneEastern Standard/Daylight Time/timeZone ntps ntp name192.43.244.18/name ntpModedirectedbroadcast/ntpMode /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameAsterisk IP/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies backupProxy/backupProxy backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxyAsterisk IP/outboundProxy outboundProxyPort5060/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures sipStack sipInviteRetx6/sipInviteRetx sipRetx10/sipRetx timerInviteExpires180/timerInviteExpires timerRegisterExpires3600/timerRegisterExpires timerRegisterDelta5/timerRegisterDelta timerKeepAliveExpires120/timerKeepAliveExpires timerSubscribeExpires120/timerSubscribeExpires timerSubscribeDelta5/timerSubscribeDelta timerT1500/timerT1 timerT24000/timerT2 maxRedirects70/maxRedirects remotePartyIDfalse/remotePartyID userInfoNone/userInfo /sipStack autoAnswerTimer1/autoAnswerTimer autoAnswerAltBehaviorfalse/autoAnswerAltBehavior autoAnswerOverridetrue/autoAnswerOverride transferOnhookEnabledfalse/transferOnhookEnabled enableVadfalse/enableVad preferredCodecg711ulaw/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandavt/dtmfOutofBand alwaysUsePrimeLinefalse/alwaysUsePrimeLine alwaysUsePrimeLineVoiceMailfalse/alwaysUsePrimeLineVoiceMail kpml3/kpml natEnabledtrue/natEnabled natAddressIP outside of NAT
Re: [asterisk-users] SendText
Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allow Header
On Monday 09 November 2009 15:38:54 Coco Richard wrote: i'm not sure to understand. Asterisk does support SIP INFO, so why doesn't Asterisk add the INFO Method in the 200OK Response? You must be using Asterisk 1.2. This is the only version that I could find that does not put the INFO tag into the Allow header. Asterisk 1.4 and all versions greater supply the INFO tag as standard. Given that 1.2 is in security-only fix mode now, this is not going to be changed in SVN or in any subsequent 1.2 release (if any). You're welcome to change the ALLOWED_METHODS define in the top of chan_sip.c and recompile, however. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.6 crashing
Maybe, you should take a look at 1.6.1.10-rc2 published yesterday. It includes an audiohook-memory patch which might correct the root cause of these crashes. As 1.6.1.9 is a security-only release, I don't think it should improve anything (beside security fix, of course). Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
On 10/11/09 4:19 PM, Thomas Perron wrote: Will text messages work to non-SIP enpoints using your logic/code? thank you If you mean SMS, yeah. Basically use SendText for devices which can display them (i.e. SIP/IAX phones) and Clickatel or the like for disconnected devices (i.e. SMS to mobile). If you wanted to extend it you could also use the Jabber functions to send to instant messaging clients. Here at the offices we basically do the following: SMS Messages for urgent notifications, payments received, support requests. Jabber Messages for incoming support call details, long Post Dial Delay warnings, congestion warnings. MRTG displaying IAX2 and SIP peer response times. Custom graphs to display inter country links. We use a system of circles around an international link. Each of our servers gets a circle. The larger the circle, the higher the delay, and if the host is unreachable the circle goes red. That way you can see from a quick glance if an international link is totally down (lots of red circles), a problem for one of our servers (one red circle), or if one of our servers is having trouble connecting to all remote links (one red circle on each link). We do the same circles for a couple of key customers to make sure their systems are always connected to multiple of our exchanges. Oh, the other thing we display on the dashboard is our Jabber statuses, and the number of tickets open in any of our support queues, and who they are assigned to. That way if someone is getting overloaded with support requests you can move jobs to another staff member. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
