Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
On 24 Oct 2008, at 03:57, David Gibbons wrote: Dare I ask why you want to do this? Cheaper than buying an AIM-CUE? And certainly more flexible. ___ -- 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] Asterisk and Cisco Call Manager Express (CME)
Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that's a different story. And I dare say that it does sound like a fun project to take on. Dave -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2008 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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] Asterisk and Cisco Call Manager Express (CME)
It's definitely just for fun, I wouldn't think to try to implement such as setup for a client unless I were really comfortable with the setup! On Fri, Oct 24, 2008 at 8:36 AM, David Gibbons [EMAIL PROTECTED] wrote: Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that's a different story. And I dare say that it does sound like a fun project to take on. Dave -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2008 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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 Cisco Call Manager Express (CME)
I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- 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] Asterisk and Cisco Call Manager Express (CME)
Dare I ask why you want to do this? Dave On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote: I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- 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
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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 Cisco Call Manager
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk and Cisco Call Manager
try with
type=peer
good luck
Edgar
On Tue, 2005-04-26 at 13:08 +0200, Alessio Focardi wrote:
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
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RE: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call gateway with the following configuration: interface Loopback79 description ALT-VoIP-Gateway ip address 10.xxx 255.255.255.255 h323-gateway voip interface h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 301 h323-gateway voip bind srcaddr 10.xxx The structure is Sip-phone à SIP à Asterisk as call-manager (extension 399) à H.323 à cisco gatekeeper (extension ) à H.323 à cisco gateway (extension 302) à E1 PSTN Iif I dial now with the Sip-phone: 302 [PSTN number (handy number, .)] I should be able to telephone the the PSTN of the gateway with the extension 302. It works within cisco devices perfectly but not with asterisk. Can you tell me your experiences and practices?? Thanks a lot!! Mario From: Mario Spendier [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 24. März 2005 13:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN Hi all, Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based. -- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi all, Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based. -- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Cisco Call Manager
Dear All : We need to use the Conference Room Capability from Asterisk to use it with our IPT Solution which based on Cisco Call Manager.. Also we need to use most of Asterisk features in our IPT Network .. How can I do this ? Any help will be grateful .. Mohamed Farid ,, Telecommunication Security Administrator ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco Call Manager.
Mind sharing how you got asterisk working with callmanager as an h323 gateway? I have configured callmanager with the ip address of my asterisk server and have setup asterisk with h323 and routed a call pattern to asterisk box but im not getting anything at the asterisk end. i have loaded the chan_h323.so module im guessing my h323.conf needs work. On Monday 09 August 2004 13:39, Andres Junge wrote: I'm in the process of doing the same thing. My approach is to declare asterisk as h323 gateway for the Cisco Call Manager, then define a route pattern to call asterisk. The strange thing that i'm dealing with now is, that the inbound RTP stream is going from the phone directly to asterisk and asterisk is sending the outbound RTP stream to asterisk. I don't know if this is a problem in asterisk or in the call manager. Salu2 Andrés Gurdeep Singh Bagga Guru escribió: Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chad Whitten Network Administrator neXband Communications [EMAIL PROTECTED] 601-944-4801 Phone 601-944-4803 Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco Call Manager.
No problem. First you need to know if the problem is that the CallManager is not sending anything or if Asterisk is not handling the conection. You can use tcpdump or ethereal for that. Salu2 Andrés Chad Whitten escribió: Mind sharing how you got asterisk working with callmanager as an h323 gateway? I have configured callmanager with the ip address of my asterisk server and have setup asterisk with h323 and routed a call pattern to asterisk box but im not getting anything at the asterisk end. i have loaded the chan_h323.so module im guessing my h323.conf needs work. On Monday 09 August 2004 13:39, Andres Junge wrote: I'm in the process of doing the same thing. My approach is to declare asterisk as h323 gateway for the Cisco Call Manager, then define a route pattern to call asterisk. The strange thing that i'm dealing with now is, that the inbound RTP stream is going from the phone directly to asterisk and asterisk is sending the outbound RTP stream to asterisk. I don't know if this is a problem in asterisk or in the call manager. Salu2 Andrés Gurdeep Singh Bagga Guru escribió: Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco Call Manager.
I'm in the process of doing the same thing. My approach is to declare asterisk as h323 gateway for the Cisco Call Manager, then define a route pattern to call asterisk. The strange thing that i'm dealing with now is, that the inbound RTP stream is going from the phone directly to asterisk and asterisk is sending the outbound RTP stream to asterisk. I don't know if this is a problem in asterisk or in the call manager. Salu2 Andrés Gurdeep Singh Bagga Guru escribió: Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco Call Manager.
Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and cisco call manager via h.323
On Tue, 16 Dec 2003, Pavel Zheltouhov wrote: Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. [...] Tested with latest cvs asterisk. Maybe asterisk h.323 channel driver not correctly parse h.323 messages. Yes, it's a known bug in chan_h323: Asterisk's RTP stack will blindly dump the RTP stream to CCM instead of the phone's IP address, which CCM correctly told us during H.245 setup. chan_oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323 will work fine. HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and cisco call manager via h.323
Does asterisk work with CCM as gateway ?
When I trying call asterisk,I totally can't hear any sound.
When call ohphone - works good.
10.0.1.219 is CCM, 10.0.1.207 asterisk.
Trace messages here :
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [5001,]
-- Calling party number: [5001]
-- Called party name: [500]
-- Called party number: [500]
-- Executing Playback(H323/ip$10.0.1.219:2303/8,
demo-abouttotry) in new stack
-- Playing 'demo-abouttotry' (language 'en')
=*= In CreateRealTimeLogicalChannel for call 8
-- externalIpAddress: 10.0.1.207
-- externalPort: 15210
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 1
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 10.0.1.207
-- ExternalPort: 15210
-- Connection Established with 5001, 5001 [10.0.1.219]
=*= In CreateRealTimeLogicalChannel for call 8
-- externalIpAddress: 10.0.1.207
-- externalPort: 15210
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 10.0.1.219
-- remotePort: 4000
-- ExternalIpAddress: 10.0.1.207
-- ExternalPort: 15210
--
remoteIpAddress: 0.0.0.0
remotePort: 0
Looks incorrectly !
Tested with latest cvs asterisk.
Maybe asterisk h.323 channel driver not correctly parse h.323 messages.
--
Pavel Zheltouhov
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