[asterisk-users] problem with asterisk - calls where both sides cannot hear each other
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
Sorry, I forgot to add one detail. Its only happening to about 5-10% of
the calls on an average day. Most of the calls goes through properly and
both sides talk and can hear each other.
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
Okay, a bit more information..
Some more information:
the non connected problem only happens to about 5-10% of the calls, the
others go through properly.. and yes, for the rest both parties can talk
and hear each other..
asterisk version:
Asterisk 1.2.13 built by root @ [hostname] on a i686 running Linux on
2006-11-14 16:53:46 UTC
We get about 50-60 calls a day with five agents.., on busy days maybe
80-100 calls
I'm setup as a call center, people call in and their calls are routed in
from analog lines through the TDM400Ps. they are connected to agents in
India via VoIP who use PolyCom IP Phones, we're using a RingAll
strategy. we have a secure IPSec tunnel between the Canada/India via
Cisco PIX501Es..
the Asterisk server has both an public (for web interface to the logs)
and private IP (10.x.x.x) and the phones have private IPs (10.x.x.x),
the traffic between them is tunneled via Cisco PIX501s.. there isn't any
NATing going on between the Asterisk and the phones..
On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
Cisco PIX501 devices, The call center in India is running 5 agents using PolyCom phones, and we're using G729 to save bandwith. And yes, we purchused 5 licenses of G729 codec. What firmware version on the Polycoms? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
010100|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=B 010100|so |3|00|Platform: Board=2345-11300-010 A 010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103, Subnet Mask=255.255.255.0 010100|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 010100|so |3|00|Application, main: Label=BOOT, Version=3.1.3.0131 27-Jan-06 12:22 010100|so |3|00|Application, main: P/N=3150-11069-313 so we're using SoundPoint IP301s.. the Firmware from my understanding is 3.1.3.131 also, would an incorrect time setting on the phones cause this problem? On Tue, 2006-12-05 at 16:17 -0600, Eric ManxPower Wieling wrote: Cisco PIX501 devices, The call center in India is running 5 agents using PolyCom phones, and we're using G729 to save bandwith. And yes, we purchused 5 licenses of G729 codec. What firmware version on the Polycoms? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
Singer Wang wrote: 010100|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=B 010100|so |3|00|Platform: Board=2345-11300-010 A 010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103, Subnet Mask=255.255.255.0 010100|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 010100|so |3|00|Application, main: Label=BOOT, Version=3.1.3.0131 27-Jan-06 12:22 010100|so |3|00|Application, main: P/N=3150-11069-313 so we're using SoundPoint IP301s.. the Firmware from my understanding is 3.1.3.131 SIP firmware would be 1.5.x, 1.6.x, or 2.0.x I recommend 1.6.7. 1.6.6 has issues, as does 2.0.x also, would an incorrect time setting on the phones cause this problem? I doubt it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
