Re: [Asterisk-Users] I connected my quicknet phonejack to the wall phone outlet and .......
It could handle two phones for sure and each acts as an extension. On Sat, 2005-09-03 at 16:20, cmisip wrote: waited for the aroma of burnt electronics. There wasn't any so I went to the next room and plugged in a telephone on the wall outlet. I picked up the handset and it rang my asterisk box. I dont have any phone service. The phone is working fine and have used it for hours listening to podcasts. And I can plug it to any phone in the house. Now I am thinking if I can plug another phone somewhere else in the house so I can have one in the kitchen too. Do you think this is asking for trouble? Assuming nothing gets fried, what happens if both handsets are picked up? Will the one phone just work as an extension. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I connected my quicknet phonejack to the wall phone outlet and .......
waited for the aroma of burnt electronics. There wasn't any so I went to the next room and plugged in a telephone on the wall outlet. I picked up the handset and it rang my asterisk box. I dont have any phone service. The phone is working fine and have used it for hours listening to podcasts. And I can plug it to any phone in the house. Now I am thinking if I can plug another phone somewhere else in the house so I can have one in the kitchen too. Do you think this is asking for trouble? Assuming nothing gets fried, what happens if both handsets are picked up? Will the one phone just work as an extension. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd
mp3 doesn't work, I recoded everything to gsm using a batch script with mplayer and sox. It took a couple of days but everything is working fine. Thanks. On Mon, 2005-08-29 at 16:00, Kris Edwards wrote: cmisip wrote: Controlplayback with the wealth of codecs supported by mplayer would be nice though as one of my future plans is sending a tv audio source through asterisk. I've done that w/ a radio, allbeit a bit convoluted setup. I was just running the live audio on an mp3 stream and then using the mp3 stream as an onhold class. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd
Thanks for the replies. It seems that controlplayback will fit my needs. I assume it supports : g723, g729, gsm, h263, ulaw, alaw, vox, wav and MP3? Does it support mp3 or must I recode to gsm? Controlplayback with the wealth of codecs supported by mplayer would be nice though as one of my future plans is sending a tv audio source through asterisk. On Sun, 2005-08-28 at 23:12, Matt Riddell wrote: trixter http://www.0xdecafbad.com wrote: On Mon, 2005-08-29 at 15:24 +1200, Matt Riddell wrote: trixter http://www.0xdecafbad.com wrote: controlplayback seems to fit if all you want is mp3s however ... Although it works with all supported formats. how many are supported? mplayer for example does at least 130 codecs making it easier to get whatever you happen to have. That was my point. :) Ok very good you win! Hehe. You always could use mencoder or other tool to convert to the desired format, but that isnt always an option (usually it is though). True, but most of the files you'd be dealing with in Asterisk would normally be created through telephony. Granted this isn't always the case though. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd
There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of media that can be piped through Asterisk. I want to be able to send a dtmf key to asterisk and have mplayer forward or rewind. I dont know much about all this stuff ( I really am not a programmer ) but I have a lot of interest in it. I use asterisk to play podcasts on my cell phone (podcasts on the road) and the forward rewind feature would be nice. Mplayer has a -slave command that allows it to read commands from stdin I think. It also has the -input parameter which allows it to read commands from a fifo. I could get the -input to work on the command line sending commands to the fifo with bash echo. However, when I am actually listening to a podcast from MP3Player in asterisk, sending a command to the fifo would cause MP3Player to respond to the command, continue playback for a short time and then exit with a No More MP3 message and a bunch of Dropping voice to exceptionally long queue Any ideas on how a forward, rewind feature can be implemented on MP3Player cmd? I have successfully used a bash script to interact with asterisk agi. My idea is to modify the app_mp3.c program to allow for a second parameter to MP3Player to pass file position value. Have app_mp3.c store the value of f-subclass into res and exit with this value whenever a key is pressed. Bash could read this value and restart the MP3Player application with a computed fileposition value as second parameter. This should achieve the result I want. Is there a simpler way? Any help or insight is appreciated. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3PLayer - rewind, forward, pause functions - feature request
mpg123 has these undocumented functions. start mpg123 with a -C argument and then pressing s will stop(pause toggle), pressing . will forward and pressing , will rewind. Can these be added to the MP3Player feature? I am using asterisk to play podcasts and this would be a nice feature. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Qos betwenn WIFI machines in LAN? oh323?
How do you do Qos between two machines when the bandwidth changes such as with WIFI? I normally get about 15 Mbit/s but this changes between 9 to 19 Mbits/s at times. Also, I use ohphone. How does one prioritize these oh323 packets or tag them for higher priority? I also have mythtv running in some machines, and this causes choppy voip when I have mythtv streaming at the same time from the same voip box. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
I got this from the voip wiki but the original script didn't seem to
work right so I fiddled with it a little bit. I am no expert so maybe
someone can look at it for errors. This is for my cable connection. So
far asterisk seems to use 1:10 while all other traffic uses 1:102. How
does one packet shape RTP?
Thanks for any help.
#!/bin/sh
TCOP=add
IPTOP=-A
if [ $1 == stop ]; then
echo Stopping...
TCOP=del
IPTOP=-D
fi
# +-+
# | root 1: |
# +-+
# |
# ++
# | class 1:1 |
# ++
# | | |
# ++ ++ ++
# |1:10| |1:20| |1:30|
# ++ ++ ++
# |
# +++
# |||
# +-+ +-+ +-+
# |1:100| |1:101| |1:102|
# +-+ +-+ +-+
# 1:10 is the class for VOIP traffic, pfifo qdisc
# 1:20 is for bulk traffic (htb, leaves use sfq)
# 1:30 is the class that interactive and TCP SYN/ACK traffic (sfq qdisc)
# 1:20 is further split up into different kinds of bulk traffic: web,
mail and
# everything else. 1:100-102 fight amongst themselves for their slice
of excess
# bandwidth, and in turn 1:10,20 and 30 then fight for any excess above
their
# minimum rates.
# which interface to throw all this on (cable)
IF=eth0
# ceil is 75% of max rate (768kbps)
# rate is 65% of max rate
RATE=500
CEIL=576
/sbin/tc qdisc ${TCOP} dev ${IF} root handle 1: htb default 102
/sbin/tc class ${TCOP} dev ${IF} parent 1: classid 1:1 htb rate
${RATE}kbit ceil ${CEIL}kbit
/sbin/tc class ${TCOP} dev ${IF} parent 1:1 classid 1:10 htb rate
64kbit ceil ${RATE}kbit prio 0
/sbin/tc class ${TCOP} dev ${IF} parent 1:1 classid 1:20 htb rate
64kbit ceil ${RATE}kbit prio 1
/sbin/tc class ${TCOP} dev ${IF} parent 1:20 classid 1:100 htb rate
${RATE}kbit prio 1
/sbin/tc class ${TCOP} dev ${IF} parent 1:20 classid 1:101 htb rate
${RATE}kbit prio 1
/sbin/tc class ${TCOP} dev ${IF} parent 1:20 classid 1:102 htb rate
${RATE}kbit prio 4
/sbin/tc qdisc ${TCOP} dev ${IF} parent 1:10 handle 10: pfifo
/sbin/tc qdisc ${TCOP} dev ${IF} parent 1:100 handle 100: sfq perturb 10
/sbin/tc qdisc ${TCOP} dev ${IF} parent 1:101 handle 101: sfq perturb 10
/sbin/tc qdisc ${TCOP} dev ${IF} parent 1:102 handle 102: sfq perturb 10
# IAX send to 1:10
/sbin/tc filter ${TCOP} dev ${IF} protocol ip parent 1:0 prio 0 u32
match ip dport 4569 0x flowid 1:10
# SIP send to 1:10
/sbin/tc filter ${TCOP} dev ${IF} protocol ip parent 1:0 prio 0 u32
match ip dport 5060 0x flowid 1:10
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
I initially used that script without modification. However, I noticed
that all traffic was going through class 1:102 regardless. Seems as if
all the children of 1:20 are set with a prio of 0 by default even if
1:20 is specifically set to prio of 2. I used
/sbin/tc -s -d class show dev eth0
to verify where the packets are going through. So I decided to set 1:10
manually to prio 0 and 1:100 and 1:101 to prio 1. I set 1:102 to prio
4(since all the rest of the traffic should be low priority).
If I run asterisk alone and make a voip iax2 call, I could see 1:1
packets incrementing as well as 1:10 which I think should be expected.
If I kill asterisk (asterisk seems to send small packets here and there
when there are no active calls) , and start a download, I see 1:1
packets incrementing, 1:20 packets incrementing and then 1:102 packets
incrementing. So I think the packets are finding their right classes.
However, there is no traffic on 1:100 and 1:101.
Like I said, I am not sure what the expected behaviour should be but I
thought all asterisk communicatios should go through 1:10.
Have you tested to see if the packets in your server are going where
they should be with the original script?
Thanks
On Sun, 2005-04-10 at 07:29, Doug Lytle wrote:
cmisip wrote:
far asterisk seems to use 1:10 while all other traffic uses 1:102. How
does one packet shape RTP?
Thanks for any help.
# +-+
# | root 1: |
# +-+
# |
# ++
# | class 1:1 |
# ++
# | | |
# ++ ++ ++
# |1:10| |1:20| |1:30|
# ++ ++ ++
# |
# +++
# |||
# +-+ +-+ +-+
# |1:100| |1:101| |1:102|
# +-+ +-+ +-+
I'm using the same script, but I found it searching Google. Yours seems
to be incomplete. My script follows:
#!/bin/sh
TCOP=add
IPTOP=-A
if [ $1 == stop ]; then
echo Stopping...
TCOP=del
IPTOP=-D
fi
# +-+
# | root 1: |
# +-+
# |
# ++
# | class 1:1 |
# ++
# | | |
# ++ ++ ++
# |1:10| |1:20| |1:30|
# ++ ++ ++
# |
# +++
# |||
# +-+ +-+ +-+
# |1:100| |1:101| |1:102|
# +-+ +-+ +-+
# 1:10 is the class for VOIP traffic, pfifo qdisc
# 1:20 is for bulk traffic (htb, leaves use sfq)
# 1:30 is the class that interactive and TCP SYN/ACK traffic (sfq qdisc)
# 1:20 is further split up into different kinds of bulk traffic: web,
mail and
# everything else. 1:100-102 fight amongst themselves for their slice
of excess
# bandwidth, and in turn 1:10,20 and 30 then fight for any excess above
their
# minimum rates.
# which interface to throw all this on (DSL)
IF=eth2
# ceil is 75% of max rate (768kbps)
# rate is 65% of max rate
# we don't let it go to 100% because we don't want the DSL modem
(Pairgain MegaBit Modem 300S)
# to have a ton of packets in their buffers. *we* want to do the buffering.
RATE=576
CEIL=640
#RATE=450
#CEIL=500
tc qdisc ${TCOP} dev ${IF} root handle 1: htb default 102
tc class ${TCOP} dev ${IF} parent 1: classid 1:1 htb rate ${RATE}kbit
ceil ${CEIL}kbit
tc class ${TCOP} dev ${IF} parent 1:1 classid 1:10 htb rate 64kbit ceil
${RATE}kbit prio 1
tc class ${TCOP} dev ${IF} parent 1:1 classid 1:20 htb rate 64kbit ceil
${RATE}kbit prio 2
tc class ${TCOP} dev ${IF} parent 1:20 classid 1:100 htb rate ${RATE}kbit
tc class ${TCOP} dev ${IF} parent 1:20 classid 1:101 htb rate ${RATE}kbit
tc class ${TCOP} dev ${IF} parent 1:20 classid 1:102 htb rate ${RATE}kbit
tc qdisc ${TCOP} dev ${IF} parent 1:10 handle 10: pfifo
tc qdisc ${TCOP} dev ${IF} parent 1:100 handle 100: sfq perturb 10
tc qdisc ${TCOP} dev ${IF} parent 1:101 handle 101: sfq perturb 10
tc qdisc ${TCOP} dev ${IF} parent 1:102 handle 102: sfq perturb 10
tc filter ${TCOP} dev ${IF} parent 1:0 protocol ip prio 1 handle 1 fw
classid 1:10
tc filter ${TCOP} dev ${IF} parent 1:0 protocol ip prio 4 handle 4 fw
classid 1:100
# IAX2 prio 0.
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 -j
MARK --set-mark 0x1
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 -j RETURN
# everything else goes into lowest priority (best effort).
iptables -t mangle ${IPTOP} PREROUTING -j MARK --set-mark 0x4
iptables -t mangle ${IPTOP} OUTPUT -j MARK --set-mark 0x4
Doug
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
After some sleep and a good breakfast, I seem to be able to think more clearly. I have come upon these conclusions: 1. Qos is all about managing upload packets ( and download packets indirectly by managing upload packets). 2. The ceiling kbit actually refers to your upload speed. It is important to keep your ceiling kbit lower than the modem upload speed, otherwise a queue gets built in the modem (most likely a FIFO queue). 3. Therefore any packets that needs to be uploaded when we hit the ceiling kbit needs to be delayed in our linux queue and prioritized according to rules that we make with tc. 4. TCP needs the ACK packets to know if packets sent have been received. It will only send more data packets if it gets this ACK back. Therefore, a delay in the return ACK packets (if you are downloading, your machine should upload ACK packets back) means TCP slows down and your download does too. 5. VOIP uses udp packets, which are fire and forget packets (no ACK packets to worry about). Therefore, to optimize VOIP, we need to put its udp packets in priority 0 with a fast FIFO queue there. 6. We probably want to put ACK in the next priority 1 and then the rest of the packets in lower priorities. 7. Qos doesn't impact incoming packets except to the extent that it influences the upload ACK packets because as soon as a packet hits your interface, its already there. If the packets are destined to the LAN, then you might setup a Qos between machines too. If these assumptions are correct, then what I see on class 1:102 is actually upload ACK packets since I see activity in that class when I do a download. Of course, I could be way out on left field here. Anybody care to comment? Thanks. I will continue my research. On Sun, 2005-04-10 at 10:05, Andrew Kohlsmith wrote: On April 10, 2005 04:47 am, cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? That looks like my rc.tc script. The most up to dateA version is at http://www.mixdown.ca/~andrew/dump/rc.tc. Please note that it only tries to make things happy for IAX2. It should be fairly easy to add RTP packet detection and to throw them into the same queue. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any gsm - g7231 codec translator?
Is there such a thing so I can call fwd from ohphone using a low bandwidth codec? What program can I use to transcode the gsm sound files in asterisk into g7231 format? If there is no way to do the above, what codec do you guys recommend if one endpoint is a dialup connection (the other broadband cable). I am using quicknet phonejacks on either end. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk - ohphone
I seem to be having problems calling ohphone from asterisk. The audio
cuts out after a few seconds with the error from ohphone:
Incoming call from root [192.168.0.1] at Sat, 02 Apr 2005 22:50:55
-0500, answer call (Y/n)? Offhook - answering call
Started logical channel: sending G.711-uLaw-64k{hw} 1
Call with root [192.168.0.1] established.
Accepting call.
Started logical channel: receiving G.711-uLaw-64k{hw} 6
Onhook - ending call.
2:17.142 H323 Cleaner assert.cxx(105) PWLib
Assertion fail: Transmit media thread did not terminate, file
channels.cxx, line 680, Error=107
Abort, Core dump, Ignore?
Calls from ohphone to asterisk work well though. Any ideas what is
going on?
Thanks.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a dial in server for internet service?
Thanks for replying. My internet connection is via cable modem. I have no telephone line though. I have VOIP working with asterisk. I was wondering if this was possible : Laptop - PSTN - Asterisk - Home network services for data connection. On Sat, 2005-03-26 at 01:18, Kannaiyan Natesan wrote: It is possible with an ISDN card, (not with analog line) Zapras application does that. http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS But it is not good to interface with analog modems, since you need to write a module to handle the soft modem. As of now nothing like that yet to asterisk. -Kannaiyan. - Original Message - From: cmisip [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 26, 2005 5:56 AM Subject: [Asterisk-Users] Asterisk as a dial in server for internet service? Is this possible? I would like to access the web through my broadband at home by dialing the home network. I think this is possible with a modem and a telephone line connected to the home network. I was wondering if Asterisk would allow such a data connection. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quicknet phonejack connect to telephone line?
Sounds like a stupid question especially since the sticker on the hardware says this will probably break the hardware if you do it. However, I just moved to a new house and the telephone wiring installed is not connected to the phone company ( the wire is just sticking out of the side of the house waiting to be connected to the phone company for service, which will NEVER happen since I got voip working.) I just want to be able to plug my telephone setsomewhere away from the room where my computer is ( like in the livingroom where there is a phone jack on the wall). Maybe plug some other phones in the other rooms as an extension. Has anybody tried this? I hate to bust my phonejack card. Thanks for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on a dialup connection?
How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required unless I can use the onbord G723.1 codecs on the quicknet cards. Ohphone allows this through h323 I think but I want an asterisk solution. If not a fullblown asterisk install on my brothers machine, maybe set it up as a h323 client to mine. I am currently working on setting up one of my lan machines with ohphone to connect to my asterisk box to call FWD and such. Is this possible? Somehow asterisk must translate the codecs from whatever SIP uses to whatever ohphone uses ( I will force it to low bandwitdh g723.1). I am hoping this will work and that the Vonage interconnect will be up soon as this will be a cheap way for them to contact my sister as well. I am still an asterisk newbie so pardon me if the questions seem newbie-ish. Has anybody gone down this path? I hate to have to reinvent the wheel. Anybody have any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as a dial in server for internet service?
Is this possible? I would like to access the web through my broadband at home by dialing the home network. I think this is possible with a modem and a telephone line connected to the home network. I was wondering if Asterisk would allow such a data connection. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD to Vonage not working?
I am having trouble with this.
I can dial 1800 numbers fine
as well as FWD service numbers but not Vonage.
I can be called from ipkall and fwd and can call aixtel numbers.
I use aix2 with Fwd.
My extensions.conf for Vonage:
; vonage numbers
;
; +2431
exten = _2431XX,1,SetCallerID,${FWDCIDNAME}
exten =
_2431XX,2,Dial,IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/**${EXTEN},60,r)
exten = _2431XX,3,Playback(invalid)
exten = _2431XX,4,Hangup
exten = _2431XX,103,Bus
The output is:
-- Executing SetCallerID(Phone/phone0, Chris) in new stack
-- Executing Dial(Phone/phone0,
IAX2/FWDNUM:[EMAIL PROTECTED]/**2431XX) in new stack
-- Called FWDNUM:[EMAIL PROTECTED]/**2431XX
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/65.39.205.121:4569/3 is busy
We're hanging up IAX2/65.39.205.121:4569/3 now...
-- Hungup 'IAX2/65.39.205.121:4569/3'
== Everyone is busy/congested at this time
Exiting with DIALSTATUS=BUSY.
No application 'Bus' for extension (default, 2431XX, 103)
== Spawn extension (default, 2431XX, 103) exited non-zero on
'Phone/phone0'
-- Hungup 'Phone/phone0'
Unfortunately, I have to use XXX's instead of . on my extensions.conf
since my Phone (via Quicknet LIte) dials the next digit immediately.
Any help is appreciated.
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to [EMAIL PROTECTED] and the Call Me test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called [EMAIL PROTECTED] No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1) (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/fwdpulvercom-dd5a is ringing Unable to handle indication 3 for 'Phone/phone0' Scheduled a registration timeout # 100 Acked pending invite 102 Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=as3d6e380d;lr=on build_route: Contact hop: sip:[EMAIL PROTECTED]:5028 -- SIP/fwdpulvercom-dd5a answered Phone/phone0 No path to translate from Phone/phone0(1) to SIP/fwdpulvercom-dd5a(2) Had to drop call because I couldn't make Phone/phone0 compatible with SIP/fwdpulvercom-dd5a update_user_counter(612) - decrement outUse counter I have a Quicknet Lite ISA card. my phone.conf contains: mode=dialtone ;format=slinear format=g723.1 echocancel=medium silencesupression=yes device = /dev/phone0 my sip.conf contains: context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register = 6:[EMAIL PROTECTED] [fwdout] type=friend username=6 secret=mypasswd host=fwd.pulver.com [fwdin] type=peer host=fwd.pulver.com context=default nat=yes canreinvite=no my extensions.conf contains: [globals] CONSOLE=Phone/phone0 [default] exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = s,1,Dial(Phone/phone0) Is it possible to call FWD using the Quicknet card? Thanks for any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
