Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Dear Mitch, Thank you so much. This partly solves my problem by a great deal, as we'll send a message to the agent immediately on picking the call. As the agents are local SIP channels, I will attempt looking up the caller's name (if it exists in our database) and set it prior to entering the queue. Is there any way of informing the agent (just) before they pick up? e.g when their phone starts ringing, so that they prepare accordingly? Regards, Wilson On Sun, Aug 4, 2013 at 4:59 AM, Mitch Claborn mitch...@claborn.net wrote: We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Dear Tiago,
Thanks for your answer, but I have a few questions.
Do you use queues? We are operating a call centre with several queues,
so I don't see how we would use the Dial command. When a call comes
in, we enter the caller (depending on what options he has selected)
into a queue. Do you have any alternative method, which would involve
dialling the agent directly as you described below?
regards,
T
On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote:
Hi,
Our queue members are Local channels, thus when dialing the agent, the
dialplan will do several stuff including:
Set(CALLERID(name)=${CALLERID(name)}:Sales)
UserEvent(something,data: ${bunch-of-data-in-some-format})
Dial(SIP/final-agent-phone,timeout,A(Sales))
The UserEvent will be picked up by our client-register-ticket-stuff software
The announcement A() will be heard by the agent upon answering the call like
sales call
On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:
We do something very similar.
Use the gosub parameter of the Queue application to call a subroutine in
the dial plan when the agent answers the call.
same =n,Queue(sales,tc,,sub-QueueConnected)
[sub-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
; whatever you need to do here
same =n,Return()
See
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
Mitch
On 08/03/2013 12:45 PM, Timothy Smith wrote:
Hello Folks,
I am setting up a call center but we have few agents so one agent is
able to handle calls of different languages and different queues. For
the agent to identify the caller, I want a popup to appear as the
phone starts to ring with the caller's number, language (selected in
the IVR), Queue (sales, support etc) and any other information (e.g a
URL with parameters)
I can send this information either via netcat (to a client such as
yac) to a Windows PC but the problem is I do not know when the caller
is about to be connected to the agent, so that I run the command. If I
wasn't using queues, it would be easy because I would run the netcat
command and then dial the user's extension.
My Question is: Is there a way I can know when the caller is just
about to be connected to an agent (when the agent's SIP extension
starts ringing)?
There are these settings setinterfacevar, setqueueentryvar,
setqueuevar in queues.conf but when can I use them?
Have you guys been in this situation before? Any alternative solutions
(sending caller info to an agent)?
I am using Asterisk 11 and Windows 7 PCs for agents.
Thank you!
Kind Regards,
Wilson
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[asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Converting MP3 files to wav for Asterisk
Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk but it wasnt any better. #!/bin/bash for i in `ls $1/*mp3` do lame -a $i $i.wav mplayer -quiet -vo null -vc dummy -ao pcm:waveheader:file=$i.h.wav $i.wav sox $i.h.wav -t raw -r 8000 -s -2 -c 1 `echo $i|sed s/.mp3/.sln/` done - Any thoughts please? Regards, Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3 Crashing Asterisk
Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Its the way some files are encoded. Is there a way I can make it skip the files that can be played? I use the Playback() and Background() Applications (Not MP3Player) Has anyone experienced this before? I searched the archives but the few posts are all for way back in 2003, so they are not so helpful. I also tried converting the files to wav or sln but there is severe music quality loss. Anyone knows a relieable way of converting the files? Thank you! Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MP3 Crashing Asterisk
Thank you for the pointers. I have checked my system, I seem to have the real mpg123. see below. -- [root@ivr2 en]# mpg123 You made some mistake in program usage... let me briefly remind you: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.13.0; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes . . . See the manpage mpg123(1) or call mpg123 with --longhelp for more parameters and information. [root@ivr2 en]# ls -l /usr/bin/mpg123 ls: cannot access /usr/bin/mpg123: No such file or directory [root@ivr2 en]# which mpg123 /usr/local/bin/mpg123 [root@ivr2 en]# ls -l /usr/local/bin/mpg123 -rwxr-xr-x. 1 root root 386286 Dec 15 00:13 /usr/local/bin/mpg123 [root@ivr2 en]# I also think I installed it using yum, however, i can still install a version from sources, just to be sure. Could you please give me the exact URLwhere I can download a version that works well with asterisk? Thank alot! Tim On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 04 Feb 2011, Timothy Smith wrote: Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Some distros used to use mpg321 instead of mpg123 (early versions of which used to suffer from non-free licence restrictions, but newer versions are LGPL) and the installer created a symbolic link so it could be invoked as mpg123. This was known to cause problems for Asterisk, which preferred the original mpg123. Try running $ mpg123 with no arguments, and note the author's name which appears in the output. If you see Michael Hipp, then it really is mpg123. If you see Joe Drew then this is really mpg321. For confirmation try $ ls -l /usr/bin/mpg123 If you see a symbolic link (cyan and permissions start with lower-case l) then this is the problem. You can always build the proper mpg123 from the Source Code (if you aren't used to doing this, you may have to install the -devel versions of any packages which you have installed but the configure script thinks you haven't, is all). When you run `make install` it probably will install itself in /usr/local/bin/mpg123 . Most distros have a default path set to look in /usr/local/bin/ before looking in /usr/bin/ ; but if you really want to make sure, then you can just copy the binary over the top of the existing symbolic link; # cp /usr/local/bin/mpg123 /usr/bin/ You might need to repeat this step last if you ever re-install mpg321 from an RPM package. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MP3 Crashing Asterisk
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: (Putting everything back into the right order, and stripping out unnecessary bits, for the sake of anybody searching the archives in future.) Thanks! On Friday 04 Feb 2011, Timothy Smith wrote: On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Try running $ mpg123 with no arguments, and note the author's name which appears in the output. Thank you for the pointers. I have checked my system, I seem to have the real mpg123. see below. [root@ivr2 en]# mpg123 You made some mistake in program usage... let me briefly remind you: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.13.0; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes Hmm . That's the real mpg123 alright. [root@ivr2 en]# which mpg123 /usr/local/bin/mpg123 I also think I installed it using yum, however, i can still install a version from sources, just to be sure. Could you please give me the exact URLwhere I can download a version that works well with asterisk? If it's in /usr/local/bin/ then it almost certainly was built from Source Code. Our working installation (on Debian Lenny) is Asterisk 1.6.2.9 (built from source) with mpg123 version 1.4.3 (installed from a .deb). More tests to try: Can you listen to an mp3 file through the Asterisk server's own sound card (if it has one; if not, use the -w option to write to a .wav file, and test that by copying it to another machine which has a sound card), by invoking mpg123 from the command line? Unfortunately, I cannot as the server is in a remote location. I also have to read about crash dumps to establish which file exactly cuases the crash. I have too much debugging but I usually see [Feb 5 08:15:51] WARNING[4895] mp3/interface.c: Junk at the beginning of frame 49443303 or [Feb 5 02:14:05] WARNING[7447]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304 just before the crash. Try $ file $(which asterisk) $ file /usr/local/bin/mpg123 and make sure both are compiled for the same architecture (ELF 64-bit LSB executable or ELF 32-bit LSB executable). If one is 32-bit and the other is 64-bit, you *will* get problems. I seem to have the same versions. [root@ivr ~]# file $(which mpg123) /usr/local/bin/mpg123: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.32, not stripped [root@ivr ~]# file $(which asterisk) /usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.32, not stripped [root@ivr1 ~]# -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manuplating Queue
Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked up? Are queues the best mothod of implimenting this? Thanks very much for your help. Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manuplating Queue
Thank you Hoggins! I am going to try it out and let you know. Regards, Tim On Sat, Sep 4, 2010 at 4:02 PM, Hoggins! fucks...@wheres5.com wrote: Hello, We use call parking for this feature. It might not be the best solution, but it works quite well. We tweaked the parking values a little bit so that the parked callers don't timeout too quickly. The receptionist fills up a dynamic list that the presenter can consult, and knows which caller he picks up by dialing 701, 702, 703, etc. (call parking is at 700 in our setup). Of course, we would have loved to have a visual solution (a good receptionist console), but we don't have time to create one for our own usage, and many solutions over the web are not compatible with our Asterisk version (1.6.2.x). Hope this helps. Hoggins! Le 04/09/2010 14:42, Timothy Smith a écrit : Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked up? Are queues the best mothod of implimenting this? Thanks very much for your help. Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding reverse dial-peer and the DTMF conf that you said.
However, I have checked my side and all seems to be ok. I've also
tried changing the dtmfmode to sip-notify on the gateway (and info in
sip.conf) but no luck!
Please look at the attached and give me some pointers.
Thanks,
Tim
On Sun, May 17, 2009 at 3:44 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com wrote:
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
___
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
dial-peer voice 1 pots
destination-pattern 0T
port 0/0/1:15
forward-digits all
!
dial-peer voice 3 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
dial-peer voice 4 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
!
dial-peer voice 112 voip
destination-pattern 730732888
monitor probe icmp-ping
session protocol sipv2
session target ipv4:172.19.3.150
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
VG2# show dial-peer voice 112
VoiceOverIpPeer112
peer type = voice, system default peer = FALSE, information type =
voice,
description = `',
tag = 112, destination-pattern = `730732888',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 112, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (To) =
Incoming (From) =
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile
[asterisk-users] Re: Asterisk With Cisco Voice Router
Thank David and Neeraj for your input. Neeraj, I posted the configs in my first post, but i've also attached some extracts here. they haven't changed much. David, You're absolutely right and i think the problem could be the reverse dial-peer or DTMF configuration. I think I have the corresponding reverse dial-peer and the DTMF conf that you said. However, I have checked my side and all seems to be ok. I've also tried changing the dtmfmode to sip-notify on the gateway (and info in sip.conf) but no luck! Please look at the attached and give me some pointers. Thanks, Tim Did you ever set up that reverse dial-peer? If not, do that first. You put a three second timeout on the Read(). By any chance, is the call hanging up 3 seconds after you call? That would be expected behavior. Well, actually you give it two tries. So it should be beep three second wait beep three second wait hangup If you're actually entering numbers on your dialpad and they're not getting read, you have a misconfiguration on your DTMF. If you enable sip debugging on your asterisk side you can see exactly what's coming over the wire from the Cisco side. There are a lot of choices for DTMF on the asterisk side and the Cisco side, and they need to agree for the button presses to be encoded and passed correctly. You can pass them in-line as real audio, or you can convert them to a special dtmf sip encoding. You'll notice all those choices when you go to configure the Cisco dial-peer. My personal preference: on the Cisco dial-peer side dtmf-relay rtp-nte on the asterisk side I left the dtmf config blank, and I don't remember which default you end up with, but it worked in the default config for me. interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable ! interface Serial0/0/1:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable dial-peer voice 1 pots destination-pattern 0T port 0/0/1:15 forward-digits all ! dial-peer voice 3 pots incoming called-number . direct-inward-dial port 0/0/1:15 dial-peer voice 4 pots incoming called-number . direct-inward-dial port 0/0/1:15 ! dial-peer voice 112 voip destination-pattern 730732888 monitor probe icmp-ping session protocol sipv2 session target ipv4:172.19.3.150 session transport udp dtmf-relay rtp-nte codec g711ulaw VG2# show dial-peer voice 112 VoiceOverIpPeer112 peer type = voice, system default peer = FALSE, information type = voice, description = `', tag = 112, destination-pattern = `730732888', voice reg type = 0, corresponding tag = 0, allow watch = FALSE answer-address = `', preference=0, CLID Restriction = None CLID Network Number = `' CLID Second Number sent CLID Override RDNIS = disabled, source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 112, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited, DTMF Relay = enabled, modem transport = system, URI classes: Incoming (Request) = Incoming (To) = Incoming (From) = Destination = huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement Translation profile (Incoming): Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4 type = voip, session-target = `ipv4:172.19.3.150', technology prefix: settle-call = disabled ip media DSCP = ef, ip signaling DSCP = af31, ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41 ip video rsvp-fail DSCP = af41, UDP checksum = disabled, session-protocol = sipv2, session-transport = udp, req-qos = best-effort, acc-qos = best-effort, req-qos video = best-effort, acc-qos video = best-effort, req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0, req-qos video def bandwidth = 384, req-qos video max bandwidth = 0, dtmf-relay = rtp-nte, RTP dynamic payload type values: NTE = 101 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122 CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0, A-law=8, GSMAMR-NB=117 iLBC=116 h263+=118, h264=119 G726r16 using static
[asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands of dollars to add those to our cisco call manager 4.1 set up. I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), and also a dialpeer to forward on the router to forward calls to my asterisk. It works properly but the problem is there is NO AUDIO! I have tried to change codec but no sucess! Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Kind Regards, Wilson sh run Building configuration... Current configuration : 6356 bytes ! ! Last configuration change at 14:21:37 UTC Fri May 15 2009 by tim ! NVRAM config last updated at 10:09:04 UTC Fri May 8 2009 by tim ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname VG2 ! boot-start-marker boot-end-marker ! card type e1 0 0 logging buffered 51200 warnings enable secret 5 $1$b4b1$BKQ.iJ/maD2vIqp3kOVQzh. ! no aaa new-model network-clock-participate wic 0 dot11 syslog ! ip cef ! ! no ip domain lookup ip domain name yourdomain.com multilink bundle-name authenticated ! isdn switch-type primary-net5 voice-card 0 no dspfarm ! ! ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw ! ! ! --More-- voice class h323 1 h225 timeout setup 4 ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /\.*/ /\1/ type any national plan any national ! ! voice translation-profile unknown_national translate called 1 ! ! ! crypto pki trustpoint TP-self-signed-4193395873 enrollment selfsigned --More-- subject-name cn=IOS-Self-Signed-Certificate-4193395873 revocation-check none rsakeypair TP-self-signed-4193395873 ! ! crypto pki certificate chain TP-self-signed-4193395873 certificate self-signed 01 30820250 308201B9 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 34313933 33393538 3733301E 170D3039 30343034 30393233 31375A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D34 3139 39353837 3330819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100B1B0 40479210 07535650 C5CFB9A2 E3F2DDFF 26D3FB27 969E69B6 25693F41 6D826D77 3A2EE99D D366247D 7035E943 B82CB36E B0C924F0 24B81745 97DCA2EA 6EF0A24E 59985063 1F1B8147 869E53BD 4F2B0827 2C9EE984 08689B22 50EF459F EEE5D22A FE52A9BB 997FC7C9 A5884405 E4993E17 52755EF5 AB88CC55 3D2C495C 36D90203 010001A3 78307630 0F060355 1D130101 FF040530 030101FF 30230603 551D1104 1C301A82 184F554C 4B4C4156 47322E79 6F757264 6F6D6169 6E2E636F 6D301F06 03551D23 04183016 8014E52D FAE14645 F6A1BBBE 21D1E27F E06FC49C 8FE9301D 0603551D 0E041604 14E52DFA E14645F6 A1BBBE21 D1E27FE0 6FC49C8F E9300D06 092A8648 86F70D01 01040500 03818100 12D38764 ABB73CD2 1E4FED39 7B765AAA 5E36CD78 1A53FEC8 2036E77A 3EBCC4D2 2E220A07 E7DF88B4 A5B6166B --More-- 24E3B3B3 CA03E0B3 EE04BCF1 831E1DB1 041C5681 FF2652D3 864CC5CC 15018B5F 0F36BA07 243E6C37 44E457CB 9CD0B4FE 15243FA8 CF15DB70 4F7C9E94 227639B1 9050906C 9ADA6A9E 27647593 94849208 75545921 quit ! ! username tim privilege 15 secret 5 $1$ojU4$n/8FgI1cRf8GhjFXd3LiU0 archive log config hidekeys ! ! controller E1 0/0/0 framing NO-CRC4 pri-group timeslots 1-31 ! controller E1 0/0/1 framing NO-CRC4 pri-group timeslots 1-31 ! ! ! --More-- ! ! interface GigabitEthernet0/0 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$ no ip address duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/0.20 encapsulation dot1Q 20 no ip address ! interface GigabitEthernet0/0.30 encapsulation dot1Q 30 ip address 172.17.3.248 255.255.255.0 h323-gateway voip interface ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto --More-- media-type rj45 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable ! interface Serial0/0/1:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable ! router eigrp 1 network 172.17.3.0 0.0.0.255 no auto-summary ! ip forward-protocol nd --More-- ! ! ip http server ip http access-class 23 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thanks Steve for this tip. I have insecure=very is not yet deprecated. I have added it but still no good. I personally think the problem could be with the codecs. Any ideas? I have attached some debug info. Regards, Tim On Sat, May 16, 2009 at 3:25 PM, Steve Howes st...@geekinter.net wrote: On 16 May 2009, at 12:46, Timothy Smith wrote: blah Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by 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 VG2# VG2# VG2# VG2# VG2# May 16 12:41:40.237: ISDN Se0/0/1:15 Q931: RX - SETUP pd = 8 callref = 0x0C73 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98392 Exclusive, Channel 18 Calling Party Number i = 0x2183, '730230199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '790792888' Plan:ISDN, Type:National May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_INCOMING May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x33B3, Guid = BCBB464BB098 VG2# May 16 12:41:40.241: fb_get_reject_cause_code: ERROR cause_code NULL May 16 12:41:40.245: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.245: ISDN Se0/0/1:15 Q931: TX - CALL_PROC pd = 8 callref = 0x8C73 Channel ID i = 0xA98392 Exclusive, Channel 18 May 16 12:41:40.353: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.353: ISDN Se0/0/1:15 Q931: TX - ALERTING pd = 8 callref = 0x8C73 VG2# May 16 12:41:46.697: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret May 16 12:41:46.697: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:46.697: ISDN Se0/0/1:15 Q931: TX - CONNECT pd = 8 callref = 0x8C73 May 16 12:41:46.713: ISDN Se0/0/1:15 Q931: RX - CONNECT_ACK pd = 8 callref = 0x0C73 May 16 12:41:46.713: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_PROGRESS OULKLAVG2# May 16 12:41:46.713: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: RX - DISCONNECT pd = 8 callref = 0x0C73 Cause i = 0x8490 - Normal call clearing May 16 12:41:52.373: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_DISC May 16 12:41:52.373: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A May 16 12:41:52.373: %ISDN-6-DISCONNECT: Interface Serial0/0/1:17 disconnected from 730230199 , call lasted 5 seconds VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: TX - RELEASE pd = 8 callref = 0x8C73 May 16 12:41:52.385: ISDN Se0/0/1:15 Q931: RX - RELEASE_COMP pd = 8 callref = 0x0C73 May 16 12:41:52.385: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_CLEARED From: 730230199 sip:730230...@172.19.3.150;tag=as5f114784 To: sip:1...@172.19.4.102:32544;rinstance=e6a140ee2d1dee0f;tag=ae700477 Contact: sip:730230...@172.19.3.150 Call-ID: 7beff1bd661329c643aa69ec43628...@172.19.3.150 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.9 Content-Length: 0 --- -- SIP/100-00820520 answered SIP/172.17.3.248-007fc920 Audio is at 172.19.3.150 port 13312 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Reliably Transmitting (no NAT) to 172.17.3.248:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK501204;received=172.17.3.248 From: sip:730230...@172.17.3.248;tag=D8FE7BF8-4CA To: sip:730232...@172.19.3.150;tag=as0fb38dd9 Call-ID: 4a137712-414d11de-9606c927-51af5...@172.17.3.248 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:730232...@172.19.3.150 Content-Type: application/sdp Content-Length: 261 v=0 o=root 544232458 544232458 IN IP4 172.19.3.150 s=Asterisk PBX 1.6.0.9 c=IN IP4 172.19.3.150 t=0 0 m=audio 13312 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Packet2Packet bridging SIP/172.17.3.248-007fc920 and SIP/100-00820520 cs-intranet*CLI --- SIP read from UDP://172.17.3.248:62582 --- ACK sip:730232...@172.19.3.150:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK511A76 From: sip:730230...@172.17.3.248;tag=D8FE7BF8-4CA
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
David,
Thanks a lot for your input. I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
cs-intranet*CLI
== Using SIP RTP CoS mark 5
-- Executing [730732...@default:1]
Read(SIP/172.17.3.248-30069280, NUM,beep,4,2,3) in new stack
-- Accepting a maximum of 4 digits.
== Using SIP RTP CoS mark 5
-- Executing [730732...@default:1]
Read(SIP/172.17.3.248-30069280, NUM,beep,4,2,3) in new stack
-- Accepting a maximum of 4 digits.
cs-intranet*CLI
Thanks alot for your assistance.
On Sat, May 16, 2009 at 4:02 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com wrote:
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully?
Yes.
You have been caught by a not-very-well-documented issue with setting
up voice routing on the 3845, and probably other similar Cisco gear.
And I'm not sure how you've done your test.
This is the closest I've ever seen to a document that explains your problem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?
If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.
At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.
you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit
A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.
___
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May 16 14:18:28.640: ISDN Se0/0/1:15 Q931: RX - RELEASE_COMP pd = 8 callref =
0x0CAB
Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.640: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No
NLCB 2
May 16 14:18:28.640: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr
0x8CAB callid 0x0
May 16 14:18:28.660: ISDN Se0/0/1:15 Q931: RX - SETUP pd = 8 callref = 0x0CAC
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838D
Exclusive, Channel 13
Calling Party Number i = 0x2183, '730730199'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '730732888'
Plan:ISDN, Type:National
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_INCOMING
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x3407, Guid
= 42D2FC70B0D1
May 16 14:18:28.664: fb_get_reject_cause_code: ERROR cause_code NULL
May 16 14:18:28.668: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 14:18:28.668: ISDN Se0/0/1:15 Q931: TX - CALL_PROC pd = 8 callref =
0x8CAC
Channel ID i = 0xA9838D
Exclusive, Channel 13
May 16 14:18:28.676
Re: [asterisk-users] Asterisk + Cisco Call Manager
Hi David, We're migrating from Cisco to asterisk because cisco is expensive to maintain, besides we can achieve more with asterisk like customised IVRs etc. This being a large production environment, we can't just change over without testing thoroughly..it has to be down in phases such that theye's no downtime at all. Now, for outgoing calls, the cisco gateways are working in parallel with asterisk. Now, i'd like to completely get rid of the cisco gateways by routing incoming calls through asterisk too (to the call manager, and finally the phones). After that, they'll be satisfied and i'll start registering the phones to asterisk until everything is asterisk. It has to be a smooth transition, just fyi, we're about 200 employees. I'll appreciate any advice towards achieving this. Kind Regards, Wison On Fri, Apr 3, 2009 at 6:14 PM, David Backeberg dbackeb...@gmail.com wrote: On Thu, Apr 2, 2009 at 12:07 PM, Timothy Smith timotsm...@gmail.com wrote: In our office, we're migrating from a Cisco set up to Asterisk. What is the goal of doing this migration? Plenty of people do a blended environment with Cisco doing what Cisco does well and Asterisk doing what Asterisk does well. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I don't really have a good idea of what call manager is / does, nor why you would want to relay incoming calls from asterisk to call manager. If you're talking about reusing IVRs or other things that you built in Cisco, those are straightforward to build in Asterisk. If you really like the GUI for building IVRs I recommend trying out FreePBX (or others), which provides a GUI on top of asterisk for tasks like that. ___ -- 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] Asterisk and Call Manager
Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make the cisco IP phones register with asterisk before it's tested thoroughly and for the gateways to be completely idle, i need to route incoming calls through asterisk. Any hints on how i can achieve this (send calls to cisco call manager 4.1 from an asterisk PBX)? Thanks in advance. Timothy ___ -- 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] Asterisk + Cisco Call Manager
Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make the cisco IP phones register with asterisk before it's tested thoroughly and for the gateways to be completely idle, i need to route incoming calls through asterisk. Any hints on how i can achieve this (send calls to cisco call manager 4.1 from an asterisk PBX)? Thanks in advance. Timothy ___ -- 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] MusicOnHold from a Sound card
Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and other tutorials on the net but no success. I have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and /usr/sbin/ast-playlinein contains #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw Am running asterisk 1.4 as root on suse 11.0 and 10.2. When I'm playing a music file (using amarok), my music onhold is silent. Is there anything I can do? Any help of pointers will be appreciated. Regards, Tim ___ -- 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] Music On Hold (from a Sound card) Help
Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no success. i have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and =/usr/sbin/ast-playlinein contains #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw Am running asterisk as root. When i'm playing a music file (using amarok), my music onhold is silent. Is there anything I can do? Any help of pointers will be appreciated. Regards, Tim ___ -- 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] Problems with Analog - SIP phone conversations
Hi, Could someone please help me with this? I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user but Sip user hears. - Calling the PSTN from the Analog phone, still the analog phone can't hear but the PSTN user hears him saying hello. repeatedly. Any help appreciated? I attempted a SIP debug and this is a sample out out: -- SIP read from 192.168.209.1:48099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received= 192.168.209.253 From: asterisk sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=as7b41af2a To: Ananth sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=2bb81ff3969 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) - --- (11 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.102.10:49166 Found description format PCMU Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED][EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.102.10, port 5060 Transmitting (NAT) to 192.168.209.1:48099: ACK sip:[EMAIL PROTECTED] [EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport From: analog-phone sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=as2b73e0bc To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=2ae01fe36af Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Regards, Tim ___ -- 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] Problems with analog - SIP phone confif\gurations
Hi, I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user but Sip user hears. - Calling the PSTN from the Analog phone, still the analog phone can't hear but the PSTN user hears him saying hello. repeatedly. Any help appreciated? I attempted a SIP debug and this is a sample out out: -- SIP read from 192.168.209.1:48099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received=192.168.209.253 From: asterisk sip:[EMAIL PROTECTED];tag=as7b41af2a To: Ananth sip:[EMAIL PROTECTED];tag=2bb81ff3969 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) - --- (11 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.102.10:49166 Found description format PCMU Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.102.10, port 5060 Transmitting (NAT) to 192.168.209.1:48099: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport From: analog-phone sip:[EMAIL PROTECTED];tag=as2b73e0bc To: sip:[EMAIL PROTECTED];tag=2ae01fe36af Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Regards, Tim ___ -- 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] FXO Hangs up automatically
Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an Acer Machine On receiving an incoming call, Connected to Asterisk 1.2.17 currently running on pbx (pid = 5092) Verbosity was 16 and is now 22 -- Starting simple switch on 'Zap/4-1' Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:4502 __zt_exception: Exception on 16, channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:3687 zt_handle_event: Got event On hook(1) on channel 4 (index 0) Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:5683 ss_thread: waitfordigit returned 0... Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2493 zt_hangup: Hangup: channel: 4 index = 0, normal = 16, callwait = -1, thirdcall = -1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 4 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2933 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1523 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' pbx*CLI On Trying to make an outgoing call Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Match Found Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits for device 319 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:6267 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Nov 20 20:51:48 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Executing Dial(SIP/319-081d8e00, Zap/1/0004479086365389) in new stack Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2065 zt_call: Dialing '0004479086365389' Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2137 zt_call: Deferring dialing... -- Called 1/0752707099 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception on 17, channel 1 Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 answered SIP/319-081d8e00 Nov 20 20:51:51 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Limit Data for this call: -- - timelimit = 0 -- - play_warning = 0 -- - warning_sound = (null) Nov 20 20:51:51 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found Nov 20 20:51:51 DEBUG[6042]: chan_sip.c:3051 sip_rtp_read: Oooh, format changed to 256 The Call doesn't go through --- Out put of `lspci` . . 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface . . . --- Output of `lsmod` Module Size Used by wctdm 37184 4 . . . - Output of /proc/zaptel/1 [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) [EMAIL PROTECTED] ~]# Output of ztcfg - [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXOKS (In use) 4 WCTDM/0/3 FXOKS (In use) -- [EMAIL PROTECTED] ~]# ztcfg - Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. [EMAIL PROTECTED] ~]# My /etc/zaptel.conf [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf fxsks=1 fxoks=2-4 loadzone = us defaultzone=us [EMAIL PROTECTED] ~]# -- My /etc/asterisk/zapata.conf [EMAIL PROTECTED] ~]# cat /etc/asterisk/zapata.conf [channels] group=2 signalling=fxo_ks context=outgoing callerid=Extensions channel = 2-4 group=3 signalling=fxs_ks context=analog-incoming channel = 1 [EMAIL PROTECTED] ~]# Out put of zap show pbx*CLI zap
