Re: [asterisk-users] help with no audio
On Tue, 01 Apr 2008 13:32:28 -0400, Jared Smith wrote: > On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: > > I call into the dialplan and try to play demo-congrats and I hear > > nothing. > > > > Firewall is disabled. > > Everything is on the 192.168.1.X network for this simple configuration. > > The tftp server is giving the polycom phone the config files. > > > > Any ideas why I dont hear audio? > > Do you happen to have an unconfigured T1 card in your machine? That's > the most common problem I see for people when they get no audio at all > coming out of Asterisk. we've seen sites where just configuring the T1/E1 card alone is not enough, we'd need to plug the card with a loopback cable or connect it to a live E1 for rtp to work. any clues why this is the case ? -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ -- 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] help with no audio
Yes, some kernels don't work with ztdummy. This is discussed over and over and over again on this mailing list. Check the archives. Tzafrir Cohen wrote: > On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote: >>> On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote: >>> / I have no card in this unit at this time. >>> />/ lsmod shows ztdummy loaded. >>> / >>> Just to make sure that this is not the problem, what's the output of: >>> >>> zttest -c 3 >>> >>> -- >> When running this nothing comes back... >> It says "Opened pseduo zap interface, measuring accuracy..." >> and that is all. >> >> I am using Centos 2.6.18-53.1.14.el5 >> >> I also just tried rmmod ztdummy and then starting asterisk again and the >> audio works. >> something is wrong with ztdummy. >> >> I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; >> make; make install >> (one at a time ) I saw no errors. tail /var/log/messages after modprove >> showed no errors. > > I believe that this means nothing. modprobe does nothing if the module > is already loaded. > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] help with no audio
On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote: > > > >On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote: > > > >>/ I have no card in this unit at this time. > >/>/ lsmod shows ztdummy loaded. > >/ > >Just to make sure that this is not the problem, what's the output of: > > > > zttest -c 3 > > > >-- > When running this nothing comes back... > It says "Opened pseduo zap interface, measuring accuracy..." > and that is all. > > I am using Centos 2.6.18-53.1.14.el5 > > I also just tried rmmod ztdummy and then starting asterisk again and the > audio works. > something is wrong with ztdummy. > > I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; > make; make install > (one at a time ) I saw no errors. tail /var/log/messages after modprove > showed no errors. I believe that this means nothing. modprobe does nothing if the module is already loaded. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] help with no audio
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote: >/ I have no card in this unit at this time. />/ lsmod shows ztdummy loaded. / Just to make sure that this is not the problem, what's the output of: zttest -c 3 -- When running this nothing comes back... It says "Opened pseduo zap interface, measuring accuracy..." and that is all. I am using Centos 2.6.18-53.1.14.el5 I also just tried rmmod ztdummy and then starting asterisk again and the audio works. something is wrong with ztdummy. I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; make; make install (one at a time ) I saw no errors. tail /var/log/messages after modprove showed no errors. Now what? 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] help with no audio
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote: > I have no card in this unit at this time. > lsmod shows ztdummy loaded. Just to make sure that this is not the problem, what's the output of: zttest -c 3 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] help with no audio
Jerry Geis wrote: >> >> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: >> >/ I call into the dialplan and try to play demo-congrats and I hear >> nothing. >> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X >> network for this simple configuration. >> />/ The tftp server is giving the polycom phone the config files. >> />/ />/ Any ideas why I dont hear audio? >> / >> Do you happen to have an unconfigured T1 card in your machine? That's >> the most common problem I see for people when they get no audio at all >> coming out of Asterisk. If that's not the case, I'd turn RTP debugging >> on in the command-line and make sure RTP packets are coming and going >> from the Asterisk box. >> >> -- >> Jared Smith >> Community Relations Manager >> Digium, Inc. > > Jared, > > I have no card in this unit at this time. > lsmod shows ztdummy loaded. > > I turned on "rtp debug" and got a bunch of lines like: > Got RTP packet from 192.168.1.99:2226 > > Does that help you? > > Jerry > I have found the echo command. I modified the dialplan to use echo. I turned on rtp debug and I see packets going BOTH ways. I have looked all through the zaptel.conf (below) everything is commented out. there are no cards in my box. zapata looks the same everything commented out. I am not finding a reason for not getting audio packets sent back to the phone. Any suggestion on something to try? Jerry - # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1 # # Note: "d4" could be referred to as "sf" or "superframe" # # The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1 # # E1's may have the additional keyword "crc4" to enable CRC4 checking # # If the keyword "yellow" follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=,,, # # Where is the name of the driver (e.g. eth), is the # driver specific address (like a MAC for eth), is the number # of channels, and is a timing priority, like for a normal span. # use "0" to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # = # # Valid devices are: # # "e&m" : Channel(s) are signalled using E&M signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # "fxsls" : Channel(s) are signalled using FXS Loopstart protocol. # "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol. # "fxsks" : Channel(s) are signalled using FXS Koolstart protocol. # "fxols" : Channel(s) are signalled using FXO Loopstart protocol. # "fxogs" : Channel(s) are signalled using FXO Groundstart protocol. # "fxoks" : Channel(s) are signalled using FXO Koolstart protocol. # "sf" : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: #channel# => sf:, # rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # "unused" : No signalling is performed, each channel in the list remains idle # "clear" : Channel(s) are bundled into a single span. No conversion or # signalling is performed, and raw data is available on the master. # "indclear": Like "clear" except all channels are treated individually and # are not bundled. "bchan" is an alias for this. # "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the # bundle, and the resulting data is communicated via the master # device. # "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the # bundle and also performs incoming and outgoin
Re: [asterisk-users] help with no audio
Jerry Geis wrote: >> >> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: >> >/ I call into the dialplan and try to play demo-congrats and I hear >> nothing. >> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X >> network for this simple configuration. >> />/ The tftp server is giving the polycom phone the config files. >> />/ />/ Any ideas why I dont hear audio? >> / >> Do you happen to have an unconfigured T1 card in your machine? That's >> the most common problem I see for people when they get no audio at all >> coming out of Asterisk. If that's not the case, I'd turn RTP debugging >> on in the command-line and make sure RTP packets are coming and going >> from the Asterisk box. >> >> -- >> Jared Smith >> Community Relations Manager >> Digium, Inc. > > Jared, > > I have no card in this unit at this time. > lsmod shows ztdummy loaded. > > I turned on "rtp debug" and got a bunch of lines like: > Got RTP packet from 192.168.1.99:2226 > > Does that help you? > > Jerry > Jared, Using the "rtp debug" I noticed that when the phone has no audio all I see is: Got RTP packet ... Got RTP packet... There are no Sent RTP packets.. under normal cases there is One sent and one Got: Got RTP packet... Sent RTP packet... Why would asterisk not be sending RTP packets I have no hardware card in this test system. Just two polycom IP330 phones. Once in a great while I will hear audio when calling into the dialplan and playing demo-congrats. 95% of the time I hear NO audio though. I am using asterisk 1.4.18, libpri 1.4.3 and zaptel 1.4.9.2 (ztdummy is loaded) Why might asterisk NOT be sending RTP packets? 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] help with no audio
> > On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: > >/ I call into the dialplan and try to play demo-congrats and I hear nothing. > />/ > />/ Firewall is disabled. > />/ Everything is on the 192.168.1.X network for this simple configuration. > />/ The tftp server is giving the polycom phone the config files. > />/ > />/ Any ideas why I dont hear audio? > / > Do you happen to have an unconfigured T1 card in your machine? That's > the most common problem I see for people when they get no audio at all > coming out of Asterisk. If that's not the case, I'd turn RTP debugging > on in the command-line and make sure RTP packets are coming and going > from the Asterisk box. > > -- > Jared Smith > Community Relations Manager > Digium, Inc. Jared, I have no card in this unit at this time. lsmod shows ztdummy loaded. I turned on "rtp debug" and got a bunch of lines like: Got RTP packet from 192.168.1.99:2226 Does that help you? 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] help with no audio
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote: > I call into the dialplan and try to play demo-congrats and I hear nothing. > > Firewall is disabled. > Everything is on the 192.168.1.X network for this simple configuration. > The tftp server is giving the polycom phone the config files. > > Any ideas why I dont hear audio? Do you happen to have an unconfigured T1 card in your machine? That's the most common problem I see for people when they get no audio at all coming out of Asterisk. If that's not the case, I'd turn RTP debugging on in the command-line and make sure RTP packets are coming and going from the Asterisk box. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] help with no audio
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play demo-congrats and I hear nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.
Any ideas why I dont hear audio?
Jerry
---
Use 'exit' when done
Asterisk 1.4.18, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
== Parsing '/etc/asterisk/asterisk.conf': Found
[0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found
[0mConnected to Asterisk 1.4.18 currently running on demobox (pid = 18129)
demobox*CLI>
Verbosity is at least 5
[Kdemobox*CLI>
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/522-051fc8f0",
"demo-congrats") in new stack
?-- Playing 'demo-congrats' (language 'en')
?
[Kdemobox*CLI>
== Spawn extension (smvoice-sip, 10, 1) exited non-zero on 'SIP/522-051fc8f0'
?
[Kdemobox*CLI> sip set debug
demobox*CLI>
SIP Debugging enabled
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" ;tag=87113650-18E1B969
To:
CSeq: 1 INVITE
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<->
?--- (14 headers 11 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - [EMAIL PROTECTED]
?
[Kdemobox*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.99:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.99;branch=z9hG4bK725666cbFE41ACBC;received=192.168.1.99
From: "522" ;tag=87113650-18E1B969
To: ;tag=as47ea6357
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ac2b96f"
Content-Length: 0
<>
?
[Kdemobox*CLI>
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method:
INVITE)
?Found user '522'
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bK725666cbFE41ACBC
From: "522" ;tag=87113650-18E1B969
To: ;tag=as47ea6357
CSeq: 1 ACK
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Max-Forwards: 70
Content-Length: 0
<->
?--- (11 headers 0 lines) ---
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.99:5060 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99;branch=z9hG4bKb6f79a4aC708EF33
From: "522" ;tag=87113650-18E1B969
To:
CSeq: 2 INVITE
Call-ID: [EMAIL PROTECTED]
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="522", realm="asterisk", nonce="5ac2b96f",
uri="sip:[EMAIL PROTECTED];user=phone",
response="9a7bd42e9bbf18fef41b63bccc83178c", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 1207070053 1207070053 IN IP4 192.168.1.99
s=Polycom IP Phone
c=IN IP4 192.168.1.99
t=0 0
m=audio 2228 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<->
?--- (15 headers 11 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.99 : 5060 (no NAT)
?Using INVITE request as basis request - [EMAIL PROTECTED]
?
[Kdemobox*CLI>
Found user '522'
?Found RTP audio format 0
?Found RTP audio format 8
?Found RTP audio format 18
?Found RTP aud
