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	<title>Comments on: Asterisk and an AS5350 SIP peer</title>
	<atom:link href="http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/feed/" rel="self" type="application/rss+xml" />
	<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/</link>
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	<lastBuildDate>Mon, 16 Apr 2012 13:37:26 +0000</lastBuildDate>
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	<item>
		<title>By: sean</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-107086</link>
		<dc:creator>sean</dc:creator>
		<pubDate>Mon, 16 Apr 2012 13:37:26 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-107086</guid>
		<description>Hi Sara,

The call queue would be implemented in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+call+queues) or whatever your PBX is. If you are using the AS5350 standalone then I think you have to do it in TCL... See http://www.cisco.com/en/US/docs/ios/voice/tcl/developer/guide/tclivrv2.html</description>
		<content:encoded><![CDATA[<p>Hi Sara,</p>
<p>The call queue would be implemented in Asterisk (see <a href="http://www.voip-info.org/wiki/view/Asterisk+call+queues" rel="nofollow">http://www.voip-info.org/wiki/view/Asterisk+call+queues</a>) or whatever your PBX is. If you are using the AS5350 standalone then I think you have to do it in TCL&#8230; See <a href="http://www.cisco.com/en/US/docs/ios/voice/tcl/developer/guide/tclivrv2.html" rel="nofollow">http://www.cisco.com/en/US/docs/ios/voice/tcl/developer/guide/tclivrv2.html</a></p>
]]></content:encoded>
	</item>
	<item>
		<title>By: sara</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-107085</link>
		<dc:creator>sara</dc:creator>
		<pubDate>Mon, 16 Apr 2012 07:39:32 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-107085</guid>
		<description>how can I have call queue in as5350 ?
is it can be configure with sip responses?</description>
		<content:encoded><![CDATA[<p>how can I have call queue in as5350 ?<br />
is it can be configure with sip responses?</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Lawrence Chew</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-103518</link>
		<dc:creator>Lawrence Chew</dc:creator>
		<pubDate>Tue, 24 Mar 2009 07:19:35 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-103518</guid>
		<description>Hi all,

How to configure SIP trunking integration between Asterik and Cisco AS5400 ?

Best regards,
Lawrence Chew</description>
		<content:encoded><![CDATA[<p>Hi all,</p>
<p>How to configure SIP trunking integration between Asterik and Cisco AS5400 ?</p>
<p>Best regards,<br />
Lawrence Chew</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: jame</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-102113</link>
		<dc:creator>jame</dc:creator>
		<pubDate>Tue, 04 Nov 2008 08:48:44 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-102113</guid>
		<description>Hi ketema,

can u share the config of ur outbound calls from Pbx asterisk to cisco as5300 setup..

i have also the same machine use but wen i try to call from pbx to cisco the called number is ringing if he answer that call both sides can&#039;t hear even f i set same codec of both ends... 
IOS Version 12.2(2)XB16

thanks james</description>
		<content:encoded><![CDATA[<p>Hi ketema,</p>
<p>can u share the config of ur outbound calls from Pbx asterisk to cisco as5300 setup..</p>
<p>i have also the same machine use but wen i try to call from pbx to cisco the called number is ringing if he answer that call both sides can&#8217;t hear even f i set same codec of both ends&#8230;<br />
IOS Version 12.2(2)XB16</p>
<p>thanks james</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Ketema</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-101075</link>
		<dc:creator>Ketema</dc:creator>
		<pubDate>Thu, 09 Oct 2008 07:11:05 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-101075</guid>
		<description>This post is very similar to what I am trying to accomplish with an AS5300.  THe problem I have though is that inbound calls from the PSTN when trying to be sent to sip-server they are rejected with SIP/2.0 488 Not acceptable here

Aterisk 1.2.12.1
IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE SOFTWARE (fc5)

I know this thread is old but has anyone see this issue ?  Outbound calls from PBX to cisco work fine.</description>
		<content:encoded><![CDATA[<p>This post is very similar to what I am trying to accomplish with an AS5300.  THe problem I have though is that inbound calls from the PSTN when trying to be sent to sip-server they are rejected with SIP/2.0 488 Not acceptable here</p>
<p>Aterisk 1.2.12.1<br />
IOS &#8482; 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE SOFTWARE (fc5)</p>
<p>I know this thread is old but has anyone see this issue ?  Outbound calls from PBX to cisco work fine.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Eusebio López</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-75197</link>
		<dc:creator>Eusebio López</dc:creator>
		<pubDate>Mon, 28 Jan 2008 09:55:21 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-75197</guid>
		<description>Hi,

I am using a Cisco AS5350 as VOIP gateway and I have a problem. 80% of incoming calls come up with the number of head, and not with the telephone number assigned to each customer in our Asterisk. Where can be the problem? I am sure that problem is configuration AS, but not it can be.

Thanks.

AS configuration is as follows:

Current configuration : 10675 bytes
!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
boot-start-marker
boot system flash c5350-is-mz.123-11.T9.bin
no boot startup-test
boot-end-marker
!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
no aaa new-model
ip subnet-zero
!
!
no ip cef
!
!
isdn switch-type primary-net5
!
!
voice service voip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 modem passthrough nse codec g711alaw
 sip
!
!
voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711alaw
!
!
!
!
controller E1 3/0
 framing NO-CRC4
 pri-group timeslots 1-31
!
controller E1 3/1
 framing NO-CRC4
 pri-group timeslots 1-31
!
controller E1 3/2
!
controller E1 3/3
!
controller E1 3/4
!
controller E1 3/5
!
controller E1 3/6
!
controller E1 3/7
!
!
interface FastEthernet0/0
 ip address 192.168.100.253 255.255.255.0 secondary
 ip address 192.168.100.2 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0
 no ip address
 shutdown
 clockrate 2000000
!
interface Serial3/0
 no ip address
 shutdown
!
interface Serial0/1
 no ip address
 shutdown
 clockrate 2000000
!
interface Serial3/0:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice modem
 no cdp enable
!
interface Serial3/1:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice modem
 no cdp enable
!
interface Async1/00
 no ip address
!
interface Async1/01
 no ip address
!
(…)
!
interface Async2/106
 no ip address
!
interface Async2/107
 no ip address
!
interface Group-Async0
 no ip address
 no group-range
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.100.3
no ip http server
!
!
!
control-plane
!
!
!
voice-port 3/0:D
!
voice-port 3/1:D
!
!
!
dial-peer voice 200 voip
 destination-pattern .
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 1000 pots
 application session
 destination-pattern .
 translate-outgoing called 1
 direct-inward-dial
 port 3/0:D
 forward-digits all
!
dial-peer voice 1001 pots
 application session
 destination-pattern .
 translate-outgoing called 1
 direct-inward-dial
 port 3/1:D
 forward-digits all
!
!
sip-ua
 sip-server ipv4:10.10.0.54
!
ss7 mtp2-variant Bellcore 0
ss7 mtp2-variant Bellcore 1
ss7 mtp2-variant Bellcore 2
ss7 mtp2-variant Bellcore 3
!
line con 0
line aux 0
line vty 0 4
  login local
line 1/00 2/107
 modem InOut
!
scheduler allocate 10000 400
ntp clock-period 17179943
ntp server 202.182.192.94
end</description>
		<content:encoded><![CDATA[<p>Hi,</p>
<p>I am using a Cisco AS5350 as VOIP gateway and I have a problem. 80% of incoming calls come up with the number of head, and not with the telephone number assigned to each customer in our Asterisk. Where can be the problem? I am sure that problem is configuration AS, but not it can be.</p>
<p>Thanks.</p>
<p>AS configuration is as follows:</p>
<p>Current configuration : 10675 bytes<br />
!<br />
version 12.3<br />
service timestamps debug datetime msec<br />
service timestamps log datetime msec<br />
service password-encryption<br />
!<br />
boot-start-marker<br />
boot system flash c5350-is-mz.123-11.T9.bin<br />
no boot startup-test<br />
boot-end-marker<br />
!<br />
!<br />
!<br />
resource-pool disable<br />
spe default-firmware spe-firmware-1<br />
no aaa new-model<br />
ip subnet-zero<br />
!<br />
!<br />
no ip cef<br />
!<br />
!<br />
isdn switch-type primary-net5<br />
!<br />
!<br />
voice service voip<br />
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco<br />
 modem passthrough nse codec g711alaw<br />
 sip<br />
!<br />
!<br />
voice class codec 1<br />
 codec preference 1 g729r8<br />
 codec preference 2 g711alaw<br />
!<br />
!<br />
!<br />
!<br />
controller E1 3/0<br />
 framing NO-CRC4<br />
 pri-group timeslots 1-31<br />
!<br />
controller E1 3/1<br />
 framing NO-CRC4<br />
 pri-group timeslots 1-31<br />
!<br />
controller E1 3/2<br />
!<br />
controller E1 3/3<br />
!<br />
controller E1 3/4<br />
!<br />
controller E1 3/5<br />
!<br />
controller E1 3/6<br />
!<br />
controller E1 3/7<br />
!<br />
!<br />
interface FastEthernet0/0<br />
 ip address 192.168.100.253 255.255.255.0 secondary<br />
 ip address 192.168.100.2 255.255.255.0<br />
 duplex auto<br />
 speed auto<br />
!<br />
interface FastEthernet0/1<br />
 no ip address<br />
 shutdown<br />
 duplex auto<br />
 speed auto<br />
!<br />
interface Serial0/0<br />
 no ip address<br />
 shutdown<br />
 clockrate 2000000<br />
!<br />
interface Serial3/0<br />
 no ip address<br />
 shutdown<br />
!<br />
interface Serial0/1<br />
 no ip address<br />
 shutdown<br />
 clockrate 2000000<br />
!<br />
interface Serial3/0:15<br />
 no ip address<br />
 isdn switch-type primary-net5<br />
 isdn incoming-voice modem<br />
 no cdp enable<br />
!<br />
interface Serial3/1:15<br />
 no ip address<br />
 isdn switch-type primary-net5<br />
 isdn incoming-voice modem<br />
 no cdp enable<br />
!<br />
interface Async1/00<br />
 no ip address<br />
!<br />
interface Async1/01<br />
 no ip address<br />
!<br />
(…)<br />
!<br />
interface Async2/106<br />
 no ip address<br />
!<br />
interface Async2/107<br />
 no ip address<br />
!<br />
interface Group-Async0<br />
 no ip address<br />
 no group-range<br />
!<br />
!<br />
ip classless<br />
ip route 0.0.0.0 0.0.0.0 192.168.100.3<br />
no ip http server<br />
!<br />
!<br />
!<br />
control-plane<br />
!<br />
!<br />
!<br />
voice-port 3/0:D<br />
!<br />
voice-port 3/1:D<br />
!<br />
!<br />
!<br />
dial-peer voice 200 voip<br />
 destination-pattern .<br />
 voice-class codec 1<br />
 session protocol sipv2<br />
 session target sip-server<br />
 dtmf-relay rtp-nte<br />
 no vad<br />
!<br />
dial-peer voice 1000 pots<br />
 application session<br />
 destination-pattern .<br />
 translate-outgoing called 1<br />
 direct-inward-dial<br />
 port 3/0:D<br />
 forward-digits all<br />
!<br />
dial-peer voice 1001 pots<br />
 application session<br />
 destination-pattern .<br />
 translate-outgoing called 1<br />
 direct-inward-dial<br />
 port 3/1:D<br />
 forward-digits all<br />
!<br />
!<br />
sip-ua<br />
 sip-server ipv4:10.10.0.54<br />
!<br />
ss7 mtp2-variant Bellcore 0<br />
ss7 mtp2-variant Bellcore 1<br />
ss7 mtp2-variant Bellcore 2<br />
ss7 mtp2-variant Bellcore 3<br />
!<br />
line con 0<br />
line aux 0<br />
line vty 0 4<br />
  login local<br />
line 1/00 2/107<br />
 modem InOut<br />
!<br />
scheduler allocate 10000 400<br />
ntp clock-period 17179943<br />
ntp server 202.182.192.94<br />
end</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Tassali Bakhsh</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-6462</link>
		<dc:creator>Tassali Bakhsh</dc:creator>
		<pubDate>Tue, 22 Aug 2006 15:53:38 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-6462</guid>
		<description>Hi Sean!
       You configured the cisco as5350 for outbound sip calls. What i want is to accept sip calls on as5350 and route them to a PRI (ISDN PBX).
       Any advise please.

Please leave me a number where i can reach you.</description>
		<content:encoded><![CDATA[<p>Hi Sean!<br />
       You configured the cisco as5350 for outbound sip calls. What i want is to accept sip calls on as5350 and route them to a PRI (ISDN PBX).<br />
       Any advise please.</p>
<p>Please leave me a number where i can reach you.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Nikolay</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-4008</link>
		<dc:creator>Nikolay</dc:creator>
		<pubDate>Fri, 14 Jul 2006 07:07:57 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-4008</guid>
		<description>I have AS5350. I configured it to work with faxes using T.37. It receives faxes from PSTN and send as an email attachment. But it refuses to send faxes from e-mail TIFF attachments to PSTN. But if I take an attachment I received from AS5350 and send it in e-mail – fax transmission to PSTN is O.K. Please, give me advice – what to do to make faxes work in both directions ?</description>
		<content:encoded><![CDATA[<p>I have AS5350. I configured it to work with faxes using T.37. It receives faxes from PSTN and send as an email attachment. But it refuses to send faxes from e-mail TIFF attachments to PSTN. But if I take an attachment I received from AS5350 and send it in e-mail – fax transmission to PSTN is O.K. Please, give me advice – what to do to make faxes work in both directions ?</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: James</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-1947</link>
		<dc:creator>James</dc:creator>
		<pubDate>Tue, 06 Jun 2006 15:31:59 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-1947</guid>
		<description>Is there anyway to use the AS5350 as a telephony bridge to use in conferencing instead of pushing the calls out of the other side (i.e. put the calls into a container on the server as one conference call)? Any help would be appreciated.</description>
		<content:encoded><![CDATA[<p>Is there anyway to use the AS5350 as a telephony bridge to use in conferencing instead of pushing the calls out of the other side (i.e. put the calls into a container on the server as one conference call)? Any help would be appreciated.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Webmaster tumujer.com</title>
		<link>http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/comment-page-1/#comment-240</link>
		<dc:creator>Webmaster tumujer.com</dc:creator>
		<pubDate>Mon, 03 Apr 2006 20:25:27 +0000</pubDate>
		<guid isPermaLink="false">http://ertw.com/blog2/?p=128#comment-240</guid>
		<description>Configure in the Cisco site:

voice-port 2/0
 vad
 cptone VE
 timing percentbreak 60
!
dial-peer voice 15 voip
 destination-pattern 2803
 session protocol sipv2
 session target ipv4:XXX.XXX.XXX.XXX
!
sip-ua 
sip-server ipv4:XXX.XXX.XXX.XXX</description>
		<content:encoded><![CDATA[<p>Configure in the Cisco site:</p>
<p>voice-port 2/0<br />
 vad<br />
 cptone VE<br />
 timing percentbreak 60<br />
!<br />
dial-peer voice 15 voip<br />
 destination-pattern 2803<br />
 session protocol sipv2<br />
 session target ipv4:XXX.XXX.XXX.XXX<br />
!<br />
sip-ua<br />
sip-server ipv4:XXX.XXX.XXX.XXX</p>
]]></content:encoded>
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