[asterisk-users] Guess I shoulda put a subject - sip diversionheader
Apparently, there is a SIP(diversionheader) field that fixes the problem below, but I cannot find any docs or examples of how to use it in my dialplan. Any help would be appreciated. We have a Cisco CallManager where users forward their numbers, so PSTN-PSTN calls get this error... -Greg --- SIP read from 209.253.136.204:5060 --- INVITE sip:[EMAIL PROTECTED];transport=UDP SIP/2.0 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY Supported: timer Accept: multipart/mixed,application/media_control+xml,application/sdp Max-Forwards: 9 Min-SE: 60 Contact: sip:[EMAIL PROTECTED]:5060;transport=UDP Content-Type: application/sdp Content-Length: 500 v=0 o=BroadWorks 31324769 1 IN IP4 209.253.136.204 s=- c=IN IP4 209.253.136.204 t=0 0 m=audio 24418 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=x-cxc-sess:04c2e65cf9a2aa97-1 a=x-cxc-info:cGVlci1wdWI9NjQuMTk5LjUxLjIxMDtwZWVyLXNkcD0yMDkuMjUzLjEyOS4xNTc6MjYzMDY7 a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MTg7 a=sendrecv - --- (14 headers 17 lines) --- Sending to 209.253.136.204 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'McLeodUSA' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 209.253.136.204:24418 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw| g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.253.136.204:24418 Looking for 9723814678 in default (domain 209.33.163.37) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=UDP ns2*CLI --- Transmitting (no NAT) to 209.253.136.204:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/4693412073-08fdbf78, SIP/[EMAIL PROTECTED]) in new stack Audio is at 192.168.5.14 port 13374 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 Apr 2008 22:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 28662 28662 IN IP4 192.168.5.14 s=session c=IN IP4 192.168.5.14 t=0 0 m=audio 13374 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 --- -- Called [EMAIL PROTECTED] ns2*CLI --- SIP read from 192.168.5.10:49365 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED];tag=16863906 Date: Thu, 17 Apr 2008 22:06:54 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (9 headers 0 lines) --- ns2*CLI --- SIP read from 192.168.5.10:6060 --- INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484 From: Cell Phone TX sip:[EMAIL PROTECTED];tag=16863908 To: sip:[EMAIL PROTECTED] Date: Thu, 17 Apr 2008 22:06:55 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: Cell Phone TX sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Contact: sip:[EMAIL PROTECTED]:6060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 227 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.5.10 s=SIP Call c=IN IP4
[asterisk-users] (no subject)
Apparently, there is a SIP(diversionheader) field that fixes the problem below, but I cannot find any docs or examples of how to use it in my dialplan. Any help would be appreciated. We have a Cisco CallManager where users forward their numbers, so PSTN-PSTN calls get this error... -Greg --- SIP read from 209.253.136.204:5060 --- INVITE sip:[EMAIL PROTECTED];transport=UDP SIP/2.0 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY Supported: timer Accept: multipart/mixed,application/media_control+xml,application/sdp Max-Forwards: 9 Min-SE: 60 Contact: sip:[EMAIL PROTECTED]:5060;transport=UDP Content-Type: application/sdp Content-Length: 500 v=0 o=BroadWorks 31324769 1 IN IP4 209.253.136.204 s=- c=IN IP4 209.253.136.204 t=0 0 m=audio 24418 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=x-cxc-sess:04c2e65cf9a2aa97-1 a=x-cxc-info:cGVlci1wdWI9NjQuMTk5LjUxLjIxMDtwZWVyLXNkcD0yMDkuMjUzLjEyOS4xNTc6MjYzMDY7 a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MTg7 a=sendrecv - --- (14 headers 17 lines) --- Sending to 209.253.136.204 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'McLeodUSA' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 209.253.136.204:24418 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw| g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.253.136.204:24418 Looking for 9723814678 in default (domain 209.33.163.37) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=UDP ns2*CLI --- Transmitting (no NAT) to 209.253.136.204:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/4693412073-08fdbf78, SIP/[EMAIL PROTECTED]) in new stack Audio is at 192.168.5.14 port 13374 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 Apr 2008 22:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 28662 28662 IN IP4 192.168.5.14 s=session c=IN IP4 192.168.5.14 t=0 0 m=audio 13374 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 --- -- Called [EMAIL PROTECTED] ns2*CLI --- SIP read from 192.168.5.10:49365 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED];tag=16863906 Date: Thu, 17 Apr 2008 22:06:54 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (9 headers 0 lines) --- ns2*CLI --- SIP read from 192.168.5.10:6060 --- INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484 From: Cell Phone TX sip:[EMAIL PROTECTED];tag=16863908 To: sip:[EMAIL PROTECTED] Date: Thu, 17 Apr 2008 22:06:55 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: Cell Phone TX sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Contact: sip:[EMAIL PROTECTED]:6060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 227 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.5.10 s=SIP Call c=IN IP4
Re: [asterisk-users] Cisco 7965 SIP Firmware
On Mon, 2008-03-31 at 23:07 +0100, Razza wrote: On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote: YMMV Change to reflect your firmware (e.g. P003-07-4-xx) 8 SNIP 8 I removed the following lines: loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8 loadInformation7 model=IP Phone 7960P003-07-4-00/loadInformation7 And tried both of these: loadInformation6 model=IP Phone 7965term65.default/loadInformation6 and loadInformation6 model=IP Phone 7965SIP45.8-3-4SR1S/loadInformation6 But again I get no further than /var/log/messages showing in.tftp stops sending after XMLDefault.cnf.xml Any suggestions? For a 7965, you might try loadinformation to be 335.. I have had to match up CCM tk.prod values to match on newer phones in the past to be what cisco uses in their internal database before I could get them to work. Although, leaving those lines out completely will work as well assuming they already have the SIP firmware loaded.. -Greg ___ -- 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] estimation on phone network capacity
On Mon, 2008-03-24 at 19:52 +0100, Philipp Kempgen wrote: mark morreny schrieb: I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. E1 is not VoIP. :-) It is if provisioned for 30 channels of concattenated data :) Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent all lines are busy? We are running a support center with mostly incoming calls. Is there any rule of thumb that are typically used for this kind of estimation? How about logging how many concurrent calls you have _today_? And I'd say it depends on how much lost calls you can tolerate. Yeah, this is 100% dependent on call volume... Regards, Philipp Kempgen ___ -- 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 as XMPP component. How to use it ?
On Feb 7, 2008, at 2:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Feb 07, 2008 at 07:53:12PM +, Ben Willcox wrote: Olivier wrote: At the opposite, I think it could be useful for an Asterisk server to act as XMPP User Activity provider (ie update XEP-0108 field with on-the-phone value). Do you agree ? Is there any XMPP client supporting User Activity ? Is Asterisk capable of getting or sending such User Activity messages ? The Openfire XMPP server (http://www.igniterealtime.org/) has an asterisk plugin which uses the manager interface to send 'On the phone' status to XMPP clients. It works very well. Is that specific to that server? What would be needed to implement the same thing on a different server? (that doesn't take 200MB of memory to boot) What about support in clients? Yeah, that is bs for a server that requires a jdk and only sends 8k messages at most. .. -- 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 ___ -- 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] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- 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] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 3:43 PM, [EMAIL PROTECTED] wrote: Greg, Without STUN how are the phones able to register? I was unable to get the Grandstream phones to work at all without STUN. -John I have nat on in sip.conf and off on the phones. Works perfect for 7960/1 and 7971. When I get back home, I will login to the asterisk servers and tell you what IPs the registration requests have in them. From : Greg Oliver [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 15:15:34 -0600 On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- 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 ___ -- 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] How to get called number in featuremap
You need $dnis.
On Jan 30, 2008, at 11:08 PM, Prashant Sharma
[EMAIL PROTECTED] wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got
executed on pressing a particular DTMF key sequence.
As I want to execute my own application on pressing that key which
will use called number.
testfeature = 3,peer,AGI,StoreNumber|CalledNumber
Here I want to use called number in place of CalledNumber tag.
When I use any variable ${DIALEDPEERNUMBER} then it does not resolve
the variable in features.conf.
if i use following then it does not work.
testfeature = 3,peer,AGI,StoreNumber|${DIALEDPEERNUMBER}
*StoreNumber is my own application that stores the number.
Any idea as how I can use CalledNumber in features.conf?
Please help.
Thanks in Advance
Regards
Prashant
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Re: [asterisk-users] transcoder
Cisco routers with DSPs as ip2ip gw will do it if you want to spend a few bucks On Jan 29, 2008, at 2:36 PM, Khaled Chehab [EMAIL PROTECTED] wrote: Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 . and forward it to a media gateway .. Regards Khaled chehab * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] SMS gateway recommendation
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote: Hi Does anyone have any recommendations of an SMS gateway which you can just sign up for on a pay-as-you-go basis for testing, for use with Asterisk? Thanks Robert McNaught In and Out Bound SMS from *, or just * - SMS? If the latter, I do not know of any provider who does not have an email - SMS gateway that is already free to use (at least in the US).. That may be the easiest way to test out your ideas.. -Greg ___ --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 Cisco calling Name
On Thu, 2007-12-06 at 10:32 -0500, John Bittner wrote: The fix for this is not to use the normal Cisco IOS. Must use 12.4T version. It is a Cisco bug. I would suggest jumping to greater than 12.4.11T as they introduced all kinds of DTMF fixes there as well.. -Original Message- On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- 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 ___ --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] Cisco 7911/7941/7970/7971 Softkey XML Files
On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... -Greg ___ --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] Cisco 7911/7941/7970/7971 Softkey XML Files
On Thu, 2007-11-15 at 05:34 +0100, Patrick wrote: On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote: The Cisco Documentation states that you can modify standard and nonstandard softkey templates. They may not be xml files. I just assumed they were xml since that is what is used to configure the phone. Just bumped into some info about this: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide_chapter09186a00807a35b9.html#wp1040919 Hope this helps. Regards, Patrick That only works when you authenticate against the phone and push xml directly to the control plane. The softkey templates from CCM are stored directly in the SQL database and are NOT flat files that can be retrieved for normal phone operation. -Greg ___ --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] Cisco 7911/7941/7970/7971 Softkey XML Files
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn’t find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. I have done packet captures to prove it out from CCM 5.x and 6.0. Sorry, no xml files to accomplish it. Maybe one day they will be less of basterds?!?!?!?!? -Greg Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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] Cisco 7941 localized menus with SIP firmware
On Thu, 2007-06-28 at 14:52 +0200, Olivier wrote: 2007/6/27, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote: Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards Actually Cisco only sendx xml for certain things. It uses a modified SIP stack and it's native SCCP stack to provision button templates, softkeys, etc.. I did hours of packet captures to try and get the info, but it is embedded into the call control stack of their phones. If you read the chan_sccp code a bit, it has a few different button layout options, that are encoded in the SCCP driver and not xml files. I wish they would go to all config files, but I doubt they will... -Greg So, if you ever use a Cisco SIP Phone with an Asterisk server, it's not possible to localize menus, soft keys, and so on ? Cheers ___ Not unless someone wants to add support for it in the SIP channel, which I doubt. I would be more than willing to provide the SIP messages that a CallManager sends to accomplish it though. -Greg ___ --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] Cisco 7941 localized menus with SIP firmware
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote: Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards Actually Cisco only sendx xml for certain things. It uses a modified SIP stack and it's native SCCP stack to provision button templates, softkeys, etc.. I did hours of packet captures to try and get the info, but it is embedded into the call control stack of their phones. If you read the chan_sccp code a bit, it has a few different button layout options, that are encoded in the SCCP driver and not xml files. I wish they would go to all config files, but I doubt they will... -Greg ___ --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] North American voice BRI - Informal survey
On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote: Hi, folks: Snip Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? -Stephen- I grew up on BRI when the internet first started taking off here. All terminated into Ascend Pipeline 50 or 25 routers. Gave 2 B and dynamic 128Kb/s bandwidth. With that said, the equipment to provision BRI on a class 5 switch here is another story. If the building they are delivering to does not have the right DLC cards, etc - it is usually chaeper for them to send a DS1 and pull 2 analog channels from it, and that is why you see BRI more exxpensive. With fiber being deployed to most buildings (or at least RTs) nowadays, the line cards do not play a factor since the DLC has to already be there. At the telco I worked, it was our philosophy to put in a mux and split out analog before going BRI. Equipment was cheaper to maintain, and provisioners were not burdened with 2 channel isdn. Now we did sell a lot of DS1 and DS3 PRIs for modem service, etc ___ --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 CCM 5.x SIP trunk
On Fri, 2007-06-01 at 10:18 +0530, Vamsi Pottangi wrote: Hi Greg, Narrowed the problem ot be that of codec mismatch ;-) Damn CCM, doesn't provide proper debugs. I have another query with CCM and Asterisk integration. In CCM cluster Phones register to 1st CCM and they fallback to 2nd incase the first fails and 3rd CCM incase even 2nd fails. How can asterisk know on which CCM subscriber the phone is registered to? How to make sure that Asterisk tries all avaiable CCMs to check where the phone is registered. Is there any better way to handle this? Thanks, ~Vamsi CCM handles all of that stuff internally. You will see SIP messages from CCMs coming from all of them all the time. It is always safest to put an entry in sip.conf for all of them in the cluster so * can at least receive calls from any of them. As far as placing calls to CCM, CCM will accept it from *, but may use any of the subscribers to route the call to. Those get set in your route list/group priorities under CCM. If you do not set any priorities, CCM will generally use the publisher of the cluster. I have never had any issues as long as all cluster CCMs were in sip.conf. -Greg ___ --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] Cisco 7961G
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 - 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric Anything older than 8.0.4SR2 is asking for grief. You cannot even download older from Cisco's website anymore. Those were their CallManager transitional loads from SCCP - SIP that were riddled with bugs. -Greg ___ --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] multiple host= in sip.conf
On Wed, 2007-05-30 at 18:03 -0500, Eric ManxPower Wieling wrote: David Boyd wrote: Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? I can also say that my experience in putting DynDNS hostnames in sip.conf do not even get mapped to IP addresses at all. I have ALWAYS had to put an actual IP for it to not grab it from eth0 by default. It never errors out while reading the config file, or logs anything - I just know it never looked up the IP for me. I have not personally tried 1.4 yet, but I would (like you) wish it to look it up and create the appropriate headers instead of me relying on my firewall to re-write them. -Greg ___ --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] Asterisk and CCM 5.x SIP trunk
On Wed, 2007-05-23 at 19:53 +0530, Vamsi Pottangi wrote: Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x? Thanks, ~Vamsi You need to make a new SIP security profile for it to work with * Under System-SecurityProfile-SIP Security Profile - you can see the trunk security settings - need to be unsecure and UDP Under Device-Trunk You will set it up using the security profile. Under Device-DeviceSettings-SIP Profile You can set all the settings. Let me know if you need more info or screenshots. Email me offline if you need some. -Greg ___ --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] Microsoft's Move Into IP PBX Market
On Wed, 2007-05-16 at 13:57 -0500, Bruce Reeves wrote: How sad, cnet misspelled Polycom and Cisco didn't make the cut. Yeah, Cisco and MSoft are on BAD terms since the inking of the deal with Nortel.. MSoft got mad when they moved from Windows Server to Linux for their CallManager platform, and it has been all downhill since then. ___ --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] Get sip response code
On Wed, 2007-05-16 at 23:19 +0100, Robert Lister wrote: I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the 604 out of this lot: -- Called [EMAIL PROTECTED] -- Got SIP response 604 Does Not Exist Anywhere back from x.x.x.x == No one is available to answer at this time (1:0/0/0) -- Executing NoOp(SIP/42105-d313f470, -- DIALSTATUS is: NOANSWER) in new stack -- Executing Goto(SIP/42105-d313f470, s-NOANSWER|1) in new stack -- Executing PlayTones(SIP/42105-d313f470, Unobtainable) in new stack -- Executing Wait(SIP/42105-d313f470, 10) in new stack Or where do I need to look to find a SIP response code - DIALSTATUS mapping? Are these configurable anywhere or are they hardcoded? If I push the response code back to the handset (Cisco 7960) then it is even more unhelpful as it uses the same error message for all SIP error type response codes: Reorder but does not tell you why the call failed to set up. If it actually put the SIP response error on the display, that would be good, but it doesn't. (at least SIP 8.6 and prior software versions) In order to display the response on the handset, Cisco phones require that you have privileges and CTI control over the phones. The only un-authenticated things you can display through the phones are through the services and directories keys. Unfortunately, they are keeping that locked up since they want you to use them with their system. Hopefully they will change their minds one day. ___ --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] Double DTMF digits
On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote: I am actually getting DTMF over SIP when people call in to a clients system that is running a2billing. They are using RFC2833. If you are using a Cisco router anywhere in the loop, there is a known bug that causes rfc2833 and inband signalling to cause double DTMF. It is fixed in IOS 12.4.11T ___ --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] Dry Copper Pair
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote: On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. You are right, but that was not what I meant, in order for one to be able to provision their own T1 over a pair of copper, the line has to allow all traffic over all frequencies pass thru it. Which these lines do not, since they are simply not just one long copper pair simply cross connected. that's what dry copper is supposed to be, just a cross connect between 2 pairs out of the CO. ie not even battery, line test equipment, or anything else hanging off it at the CO. any restriction should be purely a function of the inductance/capacitance of the wire and the connections and nothing else - anything else and you didn't get dry copper in the first place. just out of curiousity - anyone ever hijack pairs and get away with it ? (do your own cross connects on the street and utilize some crossconnect all within one branch of F1 cable out of the CO ?) I've been tempted in the past, and know that at least around here I would probably get away with it for quite some time before anyone actually cared enough to investigate. Hmmm, I can see cross connecting an F1 to the F2 to your home/business, but you would have to have a friend @ the CO to make anything of use on it right? Someone has to connect it to their frame in the CO, or xconnect it to another F1 out?? If there is a telco with live dialtone on F1 unprovisioned pairs, I would be shocked (or want to move there :) ) ___ --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] Cisco 7970 SIP won't update?
On Fri, 2006-10-13 at 11:53 -0500, Tim Connolly wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? The status screen should have errors if the config file is invalid. -Greg ___ --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] Polycom HDVoice
On Fri, 2006-10-13 at 13:08 -0500, Jessee J Holmes wrote: Actually, come to think of it, I don't know who will support it. Does Asterisk support G.722? From what I know it doesn't, is it included in the 1.4 beta? Will they support it? If Asterisk doesn't support it, then the phone won't do HD anyways. So then the questions comes to, what other PBX system or service provider will support this new HD standard? Cisco's voice gateways all either support it natively, or in pass-thru mode with newer code. Their PBX has support for it already - since their conference phones are made by polycom anyhow. ___ --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] Cisco CCM - Asterisk
On Tue, 2006-10-10 at 15:16 -0700, Alyed Tzompa wrote: What I want is to transfer some calls to a Cisco extension, so think I don't need to do the upgrade to CM5. I'm I right? Alyed Yes - you are right. On your CCM, go to a phone and check the CSS of the device and the partition of the line itself. Make sure the trunk has access to that CSS and your route pattern have access to the CSS/Partitions. CCS/Partitions can be found under: RoutePlan-Class and Control RoutePlan-Route Hunt -Greg ___ --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] Cisco 7970 Unbootable After FW Upgrade
When you do a factory reset on a 41/61/70/71, it actually deletes ALL of the firmware except the bootloader from the phone. You would have to have all of the 70s firmware files that come with them in order to boot them. The term70.default.loads tells the phone what version of software to tftp. Does the phone actually try to receive the file from your tftp server? What does your tftp log say? -Greg On Mon, 2006-10-09 at 13:23 -0500, Jeremiah Millay wrote: I tried upgrading a used Cisco 7970 from the image it shipped with to SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to do a factory reset on the phone. The phone is grabbing an IP and attempting to grab my term70.default.loads file but not moving any further. The phone screen no longer shows anything. Has anyone else had the same problem? All of my other 7970s upgraded with no problems. Since our 7970s are all used I couldn't tell what image they shipped with or what the default is. I've tried grabbing a much older SCCP image version and placing that image in my tftp server hoping it would like that but still no success. Does anyone have any suggestions as to how I can at least get this phone to boot some default SCCP image? As of right now this phone is unuseable. I get the feeling that if I can figure out what the default image is for one of these I may be able to get it to boot to that. Thanks! Jeremiah ___ --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] 7940 vs. 7941
On Thu, 2006-09-28 at 07:54 -0500, Tom wrote: At 05:39 AM 9/28/2006, you wrote: Any pros / cons on getting one over the other ? I was wondering what the main differences were. New phones (7941) support 802.3af POE. Old phones only Cisco special POE. New phones don't work with old SIP images. Only new unified SIP/SCCP images. New phones have a higher resolution display. New phones have some lighted buttons. Tom The new phones also run Java for their OS, so they are quite a bit slower than the 40/60 series for menus, etc... Their graphical displays are much higher resolution then the older models as well. -Greg ___ --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] Cisco CAll Manger and H323
4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. Anything over 4.0 supports SIP trunking. -Greg On Thu, 2006-09-28 at 19:32 +0200, Yusuf wrote: Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: stopped from reciving frames from OOH323/cisco , bridging is being stopped. What is wrong? What RTP ports must I be using? thanks, yusuf ___ --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] Cisco CAll Manger and H323
On Fri, 2006-09-29 at 20:26 -0700, Dan Austin wrote: Greg wrote: 4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. Anything over 4.0 supports SIP trunking. While it is true that CCM 4.0 and up supports SIP trunking, it is not all rainbows an butterflies. The 4.X series implimentation of SIP requires a MTP, as the SCCP endpoints do not support standards based DTMF. Additionally only ulaw is supported. In 5.X the SCCP endpoints now support RFC2833, but if you have Unity or IOS gateways, switching everything to SIP is not trivial. So for some implimentations a decent H323 channel driver is still the best option for integration. That is true, but the CCM itself is an MTP as long as you start the MoH or Media Streaming services on it.. I will say that I have not tried the 323 channels in a while (about 8 months or so), but once we switched to SIP, we have had zero issues whereas before, every 2-3 weeks, * would hiccup requiring a restart.. Our CCM phones are all Skinny since we develop products for CCM - only using * for vmail, meetme, IVR, etc.. All of our remote (all Cisco) phones use SIP connected directly to *.. 5.x CCM is much more SIP capable moreso than just the phones, but Cisco has once again done it and sends all kinds of proprietary INVITES to enable some more features on their phones.. Kinda dissapointing... I love the phones, but hate the company.. -Greg ___ --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] PSTN - CCM3.2 - Asterisk CLID
On Wed, 2006-05-24 at 07:30 -0700, Gary Richardson wrote: On the route pattern configuration page, there isn't a 'redirecting number' option. The closes thing I have is Use Calling Party's External Phone Number Mask. This is a 3.2 install of callmanager. On the gateway configuration page, there is a 'Calling Party Selection' box. Changing the values in that drop down does not have any affect on the callerid. Thanks. On 5/23/06, Greg Oliver [EMAIL PROTECTED] wrote: On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote: Hey guys, When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case. Has anyone experienced this issue before? Any solutions? Sorry - you're right - is first redirect number set on outbound calls on the gateway settings page? -Greg ___ --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] CallerID
Yeah - I have tried everything - even turning it off on the other PBX for the entire system - then XO kindly just put in the last 4 and passes it on - which would normally be OK, but the other PBX I am calling accepts that as valid and therefore I still get data.. I am going to have to get XO to turn it off momentarily or ask a bill collector to call the number for me :) Thanks, Greg On Tue, 2006-05-23 at 12:55 -0400, C F wrote: It appears that the PBX sitting between Asterisk and your provider is not passing on the calling pres flags. On 5/23/06, Lee Archer [EMAIL PROTECTED] wrote: I have a problem with BT in the UK. Using setcallerpres I can change the number shown on the recipents phones to Private or unknown but no matter what I set my asterisk cid and callerpres to it still displays the base number of my PRI ddi range. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: 23 May 2006 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID You should set the presentation flags to private. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres On 5/23/06, Greg Oliver [EMAIL PROTECTED] wrote: I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. Any ideas? TIA -Greg ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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 ___ --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] PSTN - CCM3.2 - Asterisk CLID
Yeah the Unity/CCM CTIPort combo does things with JTapi to persist the info through the system.. We us * for our IVR/VMail, and CCM for our phones.. opposite setup.. -Greg On Wed, 2006-05-24 at 08:22 -0700, Gary Richardson wrote: I've made the change, but it didn't make a difference. Unity is currently acting as our IVR. Would that make any difference? Thanks. On 5/24/06, Greg Oliver [EMAIL PROTECTED] wrote: On Wed, 2006-05-24 at 07:30 -0700, Gary Richardson wrote: On the route pattern configuration page, there isn't a 'redirecting number' option. The closes thing I have is Use Calling Party's External Phone Number Mask. This is a 3.2 install of callmanager. On the gateway configuration page, there is a 'Calling Party Selection' box. Changing the values in that drop down does not have any affect on the callerid. Thanks. On 5/23/06, Greg Oliver [EMAIL PROTECTED] wrote: On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote: Hey guys, When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case. Has anyone experienced this issue before? Any solutions? Sorry - you're right - is first redirect number set on outbound calls on the gateway settings page? ___ --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] CallerID
On Tue, 2006-05-23 at 09:32 +0100, Mark Ackroyd wrote: Here in the UK on pri, setting the callerid to 0, withholds it. I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. Tried that already - the PBX the PRI is connected to fills it in when it is invalid.. -Greg ___ --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] CallerID
On Tue, 2006-05-23 at 06:27 -0400, Steve Totaro wrote: Greg Oliver wrote: I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. Any ideas? TIA -Greg On one of my T1 circuits, ten digits always appear on the other side. I can set all ten digits to zero. If I set less than ten digits then the last digits of the default ten digit string (which is our billing phone number) are overwritten with what is set. On our T3 (different provider), we can set any length of digits but I have never tried to send blank or null values. Does your other PBX send blank callerID? Is the PRI from the same provider? When you have CIDNum= do you see errors in the log that the value must not be null? Unfortunately not - if I fill in anywhere up to the 40 digit max in *, then the other PBX allows it, but anything that is not valid, it rejects and puts the main hunt number in.. I think I am kind of screwed thanks to the 800lb gorilla Cisco Gotta love 'em. -Greg ___ --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] PSTN - CCM3.2 - Asterisk CLID
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote: Hey guys, When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case. Has anyone experienced this issue before? Any solutions? Thanks. ___ --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 Yes - you probably have redirecting number set on you rroute pattern.. Disabling that will send the correct CID, but vmail will not work.. As h.323 does not support RDNIS.. -Greg ___ --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] Centos 4.3 Issues
On Mon, 2006-05-22 at 12:16 -0400, Greg Boehnlein wrote: Hello, I was wondering if anyone out there is successfully running Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two weeks that has me scratching my head and muttering strange things in the wee hours of the morning. I am going to try and be as descriptive as my brain will allow right now, but if there is something that I do not cover, please do not hesitate to ask and I'll be happy to answer. For the last 2 years, I have been running a mixture of Tao Linux and Centos (both RHEL derivatives) on our production boxes. Asterisk has run flawlessly on all installations. Last week, I updated one of our gateway boxes from Centos 4.2 (under which it ran for 6 months without issue) to the new 4.3 code. Almost immediately, we began to experience problems. Asterisk would core w/ the following: #0 0x004878ab in test_err () from /usr/lib/asterisk/modules/codec_g729a.so The segfaults would happen under very light loads, in some cases with just a single call. Kevin was able to log in to the box, and put a debugging version of codec_g729 on the box. He determined that the problem was that the values that were being returned in that routine were incorrect. I.E. something in the system was returning a non-zero value when multiplying a number by 0. Barring any other explanations, we assumed that there was a hardware issue somewhere, either in the memory, or the FPU on the CPU. So, we replaced the box w/ a brand new Dual-Core system running a Dual-Core Pentium D 920. We loaded the 32 bit version of Centos 4.3 onto the box and proceeded to start testing. BAM.. same problem.. the backtrace showed the failure in the same routine. We scratched our heads, and after many hours of trying various things (backing off the kernel to 2.6.9-22) and even moving to the new development kernel 2.6.9-34.19 (from the testing tree) we could do nothing to solve the issue. Mind you, this is the exact same behavior on two different hardware platforms running the exact same distribution. We even loaded up a third box and could reproduce the behavior on it as well. Three different boxes, one common distribution. As a test, we installed Fedora Core 5 x86_64 on the new Dual Core box and ran extensive tests overnight, simulating 96 channels doing G729 to Ulaw transcoding. The box ran completely stable. No hiccups. So, this morning, we put it back into the cluster, and it's now taking about 200 concurrent calls, doing an insane amount of transcoding and it is working just fine. Before, it would have cored in the first couple of minutes. I'm scratching my head here, because I generally have had excellent experiences with Centos. However, I have NO idea what might be the issue here. Could it be the kernel? (We tried three different ones!). Could it be the libc? Maybe it is the compiler? In any case, if anyone is having success with Centos 4.3 (32 bit), please speak up. I'd like to get to the bottom of it. I generally do not like to run Fedora on production equipment as it is generally bleeding edge. In this case, FC5 is running 2.6.16 something.. Have you tried compiling statically on CentOS 4.2 and running on 4.3? I am assuming you have made sure the dist is up to date with patches. We do not use 729, so I cannot try it out for you, but we do use CentOS. Is it only w/ SVN, or all releases of *? -Greg ___ --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
[Asterisk-Users] CallerID
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. Any ideas? TIA -Greg ___ --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] Upgrade 7960 from SCCP 3.0 to SIP 7.5
On Sun, 2006-05-21 at 14:28 +0200, Olivier Krief wrote: Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol to which you are converting (load_information for SCCP or image_version for SIP). 3. Remove any protocol configuration files that are not used for the specified protocol. Firmware versions are P003F300 (Application Load ID) and PC030300 (Boot Load ID). In TFTP root directory, the following files are present: OS79XX.TXT P003-07-5-00.bin P003-07-5-00.sbn P0S3-07-5-00.bin P0S3-07-5-00.sbn P0S3-07-5-00.loads SIPMAC Addres.cnf SIPDefault.cnf On boot, I can see in my TFTP logs the phone is served an OS79XX.TXT file which now holds P0S3-07-5-00 content. From TFTP logs, I can see my phone is then asking for P0S3-07-.bin file which doesn't exist in my TFTP directory. Next it asked for SEPMAC Addres.cnf and SEPDefault.cnf. Both files don't exist but SIPMAC Addres.cnf and SIPDefault.cnf do exist. In SIPDefault.cnf image_version=P0S3-07-5-00 is included. . When I change P0S3-07-5-00 to P003-07-5-00 in OS79XX.TXT file, it directly asked SEPMAC Addres.cnf and SEPDefault.cnf failing to ask for any .bin file. My first question is : What should be written is OS79XX.TXT if I want to upgrade to SIP ? P0S3-07-5-00 ? P0S307500 (with a symbolic link to P0S3-07-5-00 in TFTP root directory) ? P003-07-5-00 ? You have to upgrade to a new version of SCCP or older version of SIP before the bootloader on the phone will be able to handle the newer firmware. In the same Cisco page you read the info is there - you can either use an older version of SIP first, or a newer version of SCCP.. Older SIP is probably easier - 6.3 is the newest you can use to then jump to 7.x and/or 8.x.. You will need to put this in SEPDefault.cnf (not SIPDefault) image_version:P0S3-07-5-00 (whatever version you grab) That will tell it to grab the SIP firmware if it is not using OS79XX.txt - I cant remember that far back if it is still used.. Doesn't hurt to have both though... -Greg ___ --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] CCM 3.3 and Asterisk
On Mon, 2006-05-15 at 17:40 -0300, Gustavo Souza Queiroz wrote: Hello, I´m have a CCM 3.3 and Asterisk in my LAN. I need connect my Asterisk in my CCM 3.3. You can a help me? I hate to say it, but your best bet is to upgrade to CCm 4.0 and use SIP.. It is a free cisco upgrade assuming you have a valid contract. Without that as an option, I found the ooh323 channel to be the most stable of the available ones for basic call flows.. I am pretty sure it is included in the distribution, you just have to tell it to make it. -Greg ___ --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] Compare to Skype
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote: One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine What indicates that there is no fault on his Internet connection!!! He is using his notebook and Xlite, but also tried the snom 360. Skype uses iLBC codec, which has great jitter compensation. IIRC, the newer SIP channels of * are supposed to have the same capabilities, but I have not tested. I really do not like Skype (prefer FWD), but I must say, over satellite, etc, they provide quality.. All about the codec in this case.. -Greg ___ --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] callerid name inboune from PRI
On Mon, 2006-04-10 at 22:42 -0400, Andres wrote: Steven wrote: You heard wrong. We have multiple PRIs from XO and they DO NOT send caller name. We have discussed the issue with them on several ocassions. The sales people will say whatever they want, but the tech people who actually work in the switches know that caller name is not supported. I would say it depends on the market and location as well, since XO has several flavors of switches and provisioning systems and has quite the disparate network. The ALGX infrastructure they purchased is very much still in place, and did provide CNAME on PRIs for extra cost. -Greg ___ --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
[Asterisk-Users] SIP to another PBX w/ forwarding set
OK - I know this is expected behavior, but I am stuck. Transferring calls from the * IVR to another SIP PBX ringing multiple extensions simultaneously with call-forwarding set on a phone obviously goes directly to the forwarded # since that phone answers first. I need a way to make it where if it is under say 500ms that answers, that it disregards it. I am sure I can do it with AGI, but was wondering if the normal Dial() command has an option for it? I could find no docs on it - otherwise I will write an AGI. Thanks, Greg ___ --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] # IP601's with POE per Catalyst 3560G-48PS
On Thu, 2006-04-06 at 18:57 -0700, Jay Wilton wrote: Hello people, 370 Watts maximum output / 9.6 Watts/phone = 38 phones Does this logic hold water or change with line loss? Thank you, JJW All I can say is that if you oversubscribe POE devices to a cisco switch, they have the tendency to burn out the POE modules in them.. But your logic sounds right - I am surprised to see the polycoms requiring so much power. I think in real world, you would see polycoms consuming on average much less.. -Greg ___ --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] cisco 7960
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote: does one know how to program so i can have 2 lines on one sip account on that phone ? im runnign my own asterisk do i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ? Yes. ___ --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] Anybody know about Cisco VOIP routers?
On Mon, 2006-04-03 at 13:59 -0500, Doug wrote: At 22:16 3/30/2006, Bill Gibbs wrote: Use the codec command in your dial-peer. Or a voice-class so you can have multiple supported codecs. Thanks, Bill. Could you please give an example of a voice-class entry in the dial-peer file? The voice classes we use are assigned to the physical interfaces. voice class permanent 1 signal timing idle suppress-voice 1 signal timing oos timeout disabled signal keepalive disabled signal sequence oos no-action voice-port 1/0/0 voice-class permanent 1 ___ --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] Asterisk 1.2.6, VMWare, Playback/Background GSM prompts
On Tue, 2006-03-28 at 13:08 -0500, Technical Support wrote: You can't reliably run a real-time application (like asterisk) on a virtual machine. You will get better performance from an old PC than a VM on a new top-end PC. Sorry MD H, I would have to say a properly configured GSX server running on Linux will run almost any OS and outperform and old PC. So you may want to go with GSX vs. Workstation if you don't just have a PC you can put * on by itself. -Greg ___ --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] Re: gsm picocells
On Fri, 2006-03-24 at 13:07 +1100, James Harper wrote: I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? Yes, Motorola has a hybrid wi-fi SIP/ GMS/CDMA phone in testing -Greg ___ --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] Cisco POS 3-08-2
On Wed, 2006-03-22 at 09:22 -0500, Ron Joffe wrote: On Wednesday 22 March 2006 00:33, Nathan Alberti wrote: Here is a dump of the configuration options, you will see there is a few new, these are also documented on the wiki. Nathan, How did you go about obtaining the dump ? You can always telnet into the phones and do sh config -Greg ___ --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] Cisco 7970 SIP Image
On Wed, 2006-03-22 at 11:52 +0100, Paul Brown wrote: Hi, I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-) Any pointers would be appreciated http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/ip-7900ser/cmterm-7970_7971-sip.8-0-2-0.copapp=Tablebuildstatus=showC2A ___ --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] 7970 8.x firmware speeddials
On Thu, 2006-03-23 at 02:17 +, john wrote: Hi, Does anyone know how to define speeddials in XML for the 7970 sip firmware?. I've played with the SEPmac.cnf.xml file that was posted previously but can't find a way to do it. I can define them on the phone usually (seems a bit buggy) but if the phone reboots they get lost from the config. Does anyone know a way to do this? I posted it to the sccp-users list, but here goes.. line button=3 featureID2/featureID featureLabel2000/featureLabel speedDialNumber2000/speedDialNumber /line -Greg ___ --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] voip-info.... again
On Thu, 2006-03-16 at 18:39 -0500, Alexander Lopez wrote: I have offered but I don't think he (owner) id open to that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, March 16, 2006 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voip-info again Douglas Garstang wrote: Looks like voip-info is down again today. *sigh* ___ Perhaps you should contact them and coordinate some kind of mirror. I think there are plenty of us who would provide servers for RR DNS for them. -Greg ___ --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] 7970 Configs
On Fri, 2006-03-10 at 11:52 -0600, Aaron Daniel wrote:
Anyone have the 7970 xml config for sip yet?
Aaron
[EMAIL PROTECTED] ~ $ cat SEP0014A89EF5E3.cnf.xml
device xsi:type=axl:XIPPhone ctiid=203849429
uuid={96f8508b-10ef-f98c-d20d-0471777ec725}
fullConfigtrue/fullConfig
deviceProtocolSIP/deviceProtocol
sshUserId/sshUserId
sshPassword/sshPassword
devicePool uuid={a755aa55-089c-2b47-9603-c7d51b9ca4b5}
nameDallas 5.0 Beta/name
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneGreenwich Standard Time/timeZone
/dateTimeSetting
callManagerGroup
name5.0 Beta/name
tftpDefaulttrue/tftpDefault
members
member priority=0
callManager
nameccm-beta-5-1/name
descriptionCallManager 5.0 Beta Pub - 5.0.1.032/description
ports
ethernetPhonePort2000/ethernetPhonePort
sipPort5060/sipPort
securedSipPort5061/securedSipPort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeNameccm-beta-5-1/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={cd241e11-4a58-4d3d-9661-f06c912a18a3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
sipIpAddr1/sipIpAddr1
sipPort15060/sipPort1
sipIpAddr2/sipIpAddr2
sipPort25060/sipPort2
sipIpAddr3/sipIpAddr3
sipPort35060/sipPort3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
sipProfile
sipProxies
backupProxyUSECALLMANAGER/backupProxy
backupProxyPort5060/backupProxyPort
emergencyProxyUSECALLMANAGER/emergencyProxy
emergencyProxyPort5060/emergencyProxyPort
outboundProxyUSECALLMANAGER/outboundProxy
outboundProxyPort5060/outboundProxyPort
registerWithProxytrue/registerWithProxy
/sipProxies
sipCallFeatures
cnfJoinEnabledtrue/cnfJoinEnabled
callForwardURIx-cisco-serviceuri-cfwdall/callForwardURI
callPickupURIx-cisco-serviceuri-pickup/callPickupURI
callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI
callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI
meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI
abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI
rfc2543Holdfalse/rfc2543Hold
callHoldRingback2/callHoldRingback
localCfwdEnabletrue/localCfwdEnable
semiAttendedTransfertrue/semiAttendedTransfer
anonymousCallBlock2/anonymousCallBlock
callerIdBlocking2/callerIdBlocking
dndControl0/dndControl
remoteCcEnabletrue/remoteCcEnable
/sipCallFeatures
sipStack
sipInviteRetx6/sipInviteRetx
sipRetx10/sipRetx
timerInviteExpires180/timerInviteExpires
timerRegisterExpires3600/timerRegisterExpires
timerRegisterDelta5/timerRegisterDelta
timerKeepAliveExpires120/timerKeepAliveExpires
timerSubscribeExpires120/timerSubscribeExpires
timerSubscribeDelta5/timerSubscribeDelta
timerT1500/timerT1
timerT24000/timerT2
maxRedirects70/maxRedirects
remotePartyIDtrue/remotePartyID
userInfoNone/userInfo
/sipStack
autoAnswerTimer1/autoAnswerTimer
autoAnswerAltBehaviorfalse/autoAnswerAltBehavior
autoAnswerOverridetrue/autoAnswerOverride
transferOnhookEnabledfalse/transferOnhookEnabled
enableVadfalse/enableVad
preferredCodecnone/preferredCodec
dtmfAvtPayload101/dtmfAvtPayload
dtmfDbLevel3/dtmfDbLevel
dtmfOutofBandavt/dtmfOutofBand
alwaysUsePrimeLinefalse/alwaysUsePrimeLine
alwaysUsePrimeLineVoiceMailfalse/alwaysUsePrimeLineVoiceMail
kpml3/kpml
phoneLabel/phoneLabel
stutterMsgWaiting2/stutterMsgWaiting
callStatsfalse/callStats
offhookToFirstDigitTimer15000/offhookToFirstDigitTimer
silentPeriodBetweenCallWaitingBursts10/silentPeriodBetweenCallWaitingBursts
disableLocalSpeedDialConfigtrue/disableLocalSpeedDialConfig
startMediaPort16384/startMediaPort
stopMediaPort32766/stopMediaPort
sipLines
line button=1
featureID9/featureID
featureLabel/featureLabel
proxyUSECALLMANAGER/proxy
port5060/port
name3302/name
displayName3302/displayName
autoAnswer
autoAnswerEnabled2/autoAnswerEnabled
/autoAnswer
callWaiting3/callWaiting
authName/authName
sharedLinefalse/sharedLine
messageWaitingLampPolicy3/messageWaitingLampPolicy
messagesNumber/messagesNumber
ringSettingIdle4/ringSettingIdle
ringSettingActive5/ringSettingActive
contact7b452e87-4496-4762-e11f-b26751a1884b/contact
forwardCallInfoDisplay
callerNametrue/callerName
callerNumberfalse/callerNumber
redirectedNumberfalse/redirectedNumber
dialedNumbertrue/dialedNumber
/forwardCallInfoDisplay
/line
/sipLines
voipControlPort5060/voipControlPort
dscpForAudio184/dscpForAudio
ringSettingBusyStationPolicy0/ringSettingBusyStationPolicy
dialTemplate/dialTemplate
softKeyFileSK50719900-3bee-4594-bc3f-6400e1a33bf0.xml/softKeyFile
/sipProfile
commonProfile
phonePassword/phonePassword
backgroundImageAccesstrue/backgroundImageAccess
callLogBlfEnabled2/callLogBlfEnabled
/commonProfile
loadInformationSIP70.8-0-0-38S/loadInformation
vendorConfig
Re: [Asterisk-Users] 7970 Configs
If I recall when we first got the CCM5 development SIP loads, I got the
same result, but it was funny that * showed the phone as not registered.
It may well be the fact that I have not downloaded the released version.
It may be more non-CCM friendly.
I'll play with it again next week if I can borrow a 70 away from the
developers for a while.
The only thing I do not like about the 41/61/70/71 (all the java phones)
is they only allow one password for all the separate lines/proxies in
SIP mode. I may play with the config to see if it will allow more.
-Greg
BTW: If you do get it to play nice, please post the xml file for us :)
On Fri, 2006-03-10 at 13:56 -0600, Aaron Daniel wrote:
Awesome, that works, 'cept now the dialplan doesn't work lol. I've
programmed the voicemail button in, but anything I try to dial doesn't
make it past the first digit.
Aaron
Greg Oliver wrote:
On Fri, 2006-03-10 at 11:52 -0600, Aaron Daniel wrote:
Anyone have the 7970 xml config for sip yet?
Aaron
[EMAIL PROTECTED] ~ $ cat SEP0014A89EF5E3.cnf.xml
device xsi:type=axl:XIPPhone ctiid=203849429
uuid={96f8508b-10ef-f98c-d20d-0471777ec725}
fullConfigtrue/fullConfig
deviceProtocolSIP/deviceProtocol
sshUserId/sshUserId
sshPassword/sshPassword
devicePool uuid={a755aa55-089c-2b47-9603-c7d51b9ca4b5}
nameDallas 5.0 Beta/name
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneGreenwich Standard Time/timeZone
/dateTimeSetting
callManagerGroup
name5.0 Beta/name
tftpDefaulttrue/tftpDefault
members
member priority=0
callManager
nameccm-beta-5-1/name
descriptionCallManager 5.0 Beta Pub - 5.0.1.032/description
ports
ethernetPhonePort2000/ethernetPhonePort
sipPort5060/sipPort
securedSipPort5061/securedSipPort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeNameccm-beta-5-1/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={cd241e11-4a58-4d3d-9661-f06c912a18a3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
sipIpAddr1/sipIpAddr1
sipPort15060/sipPort1
sipIpAddr2/sipIpAddr2
sipPort25060/sipPort2
sipIpAddr3/sipIpAddr3
sipPort35060/sipPort3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
sipProfile
sipProxies
backupProxyUSECALLMANAGER/backupProxy
backupProxyPort5060/backupProxyPort
emergencyProxyUSECALLMANAGER/emergencyProxy
emergencyProxyPort5060/emergencyProxyPort
outboundProxyUSECALLMANAGER/outboundProxy
outboundProxyPort5060/outboundProxyPort
registerWithProxytrue/registerWithProxy
/sipProxies
sipCallFeatures
cnfJoinEnabledtrue/cnfJoinEnabled
callForwardURIx-cisco-serviceuri-cfwdall/callForwardURI
callPickupURIx-cisco-serviceuri-pickup/callPickupURI
callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI
callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI
meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI
abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI
rfc2543Holdfalse/rfc2543Hold
callHoldRingback2/callHoldRingback
localCfwdEnabletrue/localCfwdEnable
semiAttendedTransfertrue/semiAttendedTransfer
anonymousCallBlock2/anonymousCallBlock
callerIdBlocking2/callerIdBlocking
dndControl0/dndControl
remoteCcEnabletrue/remoteCcEnable
/sipCallFeatures
sipStack
sipInviteRetx6/sipInviteRetx
sipRetx10/sipRetx
timerInviteExpires180/timerInviteExpires
timerRegisterExpires3600/timerRegisterExpires
timerRegisterDelta5/timerRegisterDelta
timerKeepAliveExpires120/timerKeepAliveExpires
timerSubscribeExpires120/timerSubscribeExpires
timerSubscribeDelta5/timerSubscribeDelta
timerT1500/timerT1
timerT24000/timerT2
maxRedirects70/maxRedirects
remotePartyIDtrue/remotePartyID
userInfoNone/userInfo
/sipStack
autoAnswerTimer1/autoAnswerTimer
autoAnswerAltBehaviorfalse/autoAnswerAltBehavior
autoAnswerOverridetrue/autoAnswerOverride
transferOnhookEnabledfalse/transferOnhookEnabled
enableVadfalse/enableVad
preferredCodecnone/preferredCodec
dtmfAvtPayload101/dtmfAvtPayload
dtmfDbLevel3/dtmfDbLevel
dtmfOutofBandavt/dtmfOutofBand
alwaysUsePrimeLinefalse/alwaysUsePrimeLine
alwaysUsePrimeLineVoiceMailfalse/alwaysUsePrimeLineVoiceMail
kpml3/kpml
phoneLabel/phoneLabel
stutterMsgWaiting2/stutterMsgWaiting
callStatsfalse/callStats
offhookToFirstDigitTimer15000/offhookToFirstDigitTimer
silentPeriodBetweenCallWaitingBursts10/silentPeriodBetweenCallWaitingBursts
disableLocalSpeedDialConfigtrue/disableLocalSpeedDialConfig
startMediaPort16384/startMediaPort
stopMediaPort32766
RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg ___ --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] 7940/60 SIP 8.2
On Thu, 2006-03-09 at 13:11 -0600, Shane Young wrote: Quoting Mailing List [EMAIL PROTECTED]: I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. They support third-party partners such as Broadsoft. Broadsoft is entirely SIP - just like the channel in Asterisk. They utilize the Cisco XML features just like anyone else could though. -Greg ___ --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] NEWS: SIP Firmware Available for Cisco 7970
On Mon, 2006-03-06 at 22:11 -0500, Darren Wright wrote: InterestingI've upgraded the 7970 to SIP, but it is still saying unprovisioned. I've got a SIPMAC file, but it is still looking for the SEPMAC file... That's correct - the CCM5 loads only look for SEP files. Even when you give it one, it will not register with Asterisk. If you need a fully formatted SEPxml file, I will email you one off line for a 70. Anyone got this working yet? Nope :( -D ___ --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] NEWS: SIP Firmware Available for Cisco 7970
On Mon, 2006-03-06 at 12:38, Nabeel Jafferali wrote: I have a service contract for my 7960 but I don't see 8.x SIP firmware for it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960. I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the 7960. You have to have developer support contracts to currently get to them. -Greg ___ --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] call manager integration
On Mon, 2006-03-06 at 15:00, Jerry Geis wrote: I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? What is the output on the console with sip debug turned on? -Greg ___ --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] call manager integration
On Mon, 2006-03-06 at 15:42, Jerry Geis wrote: here is some of the output. I am no longer the to spcifically do sip debug but this is what I have. along with my sip.conf snip. The call to extension 3726 never rings. so it never gets answered. Are you sure your sip trunk and route pattern are in the same partition/CSS by chance? Without more info (AGI script and SIP debug), I really can't be much more help. Your sip.conf entry is good though. Your callmanager context from extensions.conf will help as well. -Greg ___ --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] NEWS: SIP Firmware Available for Cisco 7970
On Mon, 2006-03-06 at 15:59, Mailing List wrote: tar zxfv *.cop - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 06, 2006 4:00 PM Subject: Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970 Ok, so, we've got the 7970 SIP Firmware now, but their readme is a little sparse... Anyone have any clue as to the upgrade procedure for a non-ccm5 system? (i.e. asterisk ;)) Aaron Julien Goodwin wrote: I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in .cop files, these are actually just tarballs (.tar.gz) with a new name. The names are mangled, but relativly easy to figure out. Please note that I will not give this firmware out, nor point people to places where they may pirate it. Thanks, Julien Inside, you should have files like... P70.8-0-0-38S.loads jar70sip.8-0-0-38.sbn cnu70.3-0-1-63.sbn apps70.1-1-0-63.sbn dsp70.1-1-0-63.sbn cvm70sip.8-0-0-38.sbn You upgrade the same way you would a 40/60 leaving the .loads off of the firmware name. I have tested and have not successfully gotten any CCM5.0 SIP loads to register with asterisk though. I will try some more when I have time to do some packet captures and analyze them later in the week. -Greg ___ --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] Program Buttons on Cisco 79xx Phones
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote: Unfortunately you have to make a choice: SIP firmware - Easy to implement on *, but poor XML support SCCP firmware - poor/non-trivial asterisk support, great XML support. The newest SIP firmware (beta versions) allows the exact XML functionality as the SCCP versions. Since Cisco CME and CCM are both migrating to SIP (CME already has) in their next versions, the loads provide all the bells and whistles. We have the development loads and have every Cisco model currently running in SIP. Currenltly, I cannot register to * with them since there is no auth in them yet. For example - on a 40/60, it only allows a single global logon (not per line like current SIP firmware) and that does not even work - never tries to register with asterisk at all and I only had about an hour at work to spend on it. I can say that the numbering scheme for SIP loads (7.5, etc) has remained intact. So the probable reason there has not been a 7.6 release even with all of the 7.5 bugs is because the 8.0 version is in beta/development right now. They are gonna be awesome phones once they have all the SCCP capabilities for SIP if the decide to merge the codebase into CCM/3rd-party compatibility. -Greg ___ --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] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann [EMAIL PROTECTED] wrote: Joao Pereira wrote: Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan your data vlan switchport mode trunk switchport voice vlan your voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --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 ___ --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] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
I have never used a switchport for .1x to a PC connected through a phone. I would say it probably will not work since it bypasses the idea of .1x entirely if it does. You maybe could use it in 802.11 mode, but the phone would probably not have access until the PC auths (if it would work at all).. On Thu, 2006-03-02 at 16:51 +, Joao Pereira wrote: And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as access (with 802.1x) or trunk (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann [EMAIL PROTECTED] wrote: Joao Pereira wrote: Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan your data vlan switchport mode trunk switchport voice vlan your voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --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 ___ --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 ___ --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] wake up calls
On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: Does anyone have a way to do wake calls? Jordan Novak Communications Technician Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... ___ --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] OT- Rwanda DSL growth
On Sat, 2006-02-25 at 13:42 -0500, Dean Collins wrote: I know this is a OT but great article http://www.theregister.co.uk/2006/02/23/rwanda_terracom/ Will be interesting to see how this project goes. Hmmm - it is nice to see things like this happening, but I would have thought that 802.16 would have been the prevalent technology in these areas since CPE has been ratified finally and its reach is much more cost effective. Nonetheless, I do agree. -Greg ___ --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] OT: VoIP over bonded link
The above concern have been a major issue with telephone equipment (eg, central offices) and the telco's spend a significant amount of money burying very long rods in the ground and interconnectng them with the CO hardware using cables that are larger then 1/4 in diameter (don't remember the guage anymore). Every row of racks include the heavy ground cabling, and rack paint (etc) is often times scrapped off between racks to ensure a solid ground. They use special test equipment to actually measure the implementation. Yes - the last LEC I worked for grounded every single relay rack to the DC power plant ground with #6 cable. Might be overkill, but the cost of the cable versus a SONET shelf or DLC is definitely worth 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
Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection
Depends on the type of satellite, but generally 1500 - 3000ms. On Wed, 2006-02-01 at 18:28 +0100, Master_PE wrote: What is a normal dealy on a satelite installation? Regards, Master_PE Op 1-feb-2006, om 13:26 heeft Garth van Sittert het volgende geschreven: Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/ bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of call center. That is, we want to get a few land-lines from our telco in different countys and bridge all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP in our area that can deliver a broadband connection for anything less then an arm and a leg, so we're considering runing an * - * connection using VoIP over a low bandwidth connection (we're considering 128kbit but we might be able to go to 256kbit). The bandwidth price is not a problem for our satelite installations, we cand get acceptably priced broadband (~256kbit) so the distant *'s will have propper connections. My question: Is 128kbit a wide enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund ___ --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 From - Wed -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --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 ___ --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] SIP-H323 translation
I have found * with the ooh323 channel to be best for this. On Mon, 2006-01-30 at 15:23 +0200, [EMAIL PROTECTED] wrote: Hello, I would like to find an appropriate solution for SIP to H323 translation (vice versa would be great too!), in an environment where there's going to be 100+ concurrent calls: has anyone succesfully implemented such a translator/gateway, e.g. using Opal +OpenH323/Asterisk or anything else? Any idea of the requisites or issues that could be faced? Thank you! Tim ___ --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] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk
You can have asterisk dial your Unity vmail pilot on busy or unavailable, and have CCM use the last redirected number on the trunk to determine the called extension, or pass the $RDNIS value and digit add/strip from * to CCM. We use * in the exact opposite fashion, but should suffice either direction. On Tue, 2006-01-24 at 07:56 -0800, sys read wrote: I have my eyes on the Linksys/Sipura 941, ( SIP ), but the core problem is that you can't use SIP phones with CCM. I have a SIP trunk between asterisk and ccm. I can route calls back and forth, I just can't get the call to send to vm if no answer on the asterisk side. On 1/24/06, kevin ling [EMAIL PROTECTED] wrote: Hi, Maybe buy 7912 phone and register to CCM is another choice. or integrated CCM with asterisk voicemail system. __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sys read Sent: Tuesday, January 24, 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk Hi guys, I want to leave messages on our unity box. I have already converted a couple 7940s to SIP, but I can't give them out to our users because I don't want to have to deal with two voicemail systems. we have licenses for all our users on unity as is.we're about to buy a bunch more 7940s, but I don't want to cause they're expensive. I'd rather buy a cheaper SIP phone and have it rollover to the unity vm. On 1/23/06, Gary Richardson [EMAIL PROTECTED] wrote: You can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty good support for it. Check the voip-info.org wiki for instructions on switching the firmware. Hopefully that will take a step out of the plan -- you could completely ditch your Cisco system :) On 1/23/06, sys read [EMAIL PROTECTED] wrote: Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each, the alternative is cisco 7940s ( around $250 ) running SCCP through the CCM. at the quantities I'm talking about, $100 is significant. Does anyone have any idea how to get this done? I've tried this: exten = 123,1,Dial(SIP/sipphone,20) exten = 123,2,Dial(SIP/ccm/3040) where 3040 is our VM pilot for ccm. but all it does is take us to the main greeting. we have smartnet, but they haven't been helpful at all I called digium to see if they could help if we paid, but they said they've never heard of cisco unity help? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation
Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk
You can use setvar to append it to the extension of your vmail pilot,
and strip the vmail pilot number off on the trunk side in CCM.
eg:
SetVar(TEST=$[${EXTEN}${RDNIS}])
Dial(SIP/[EMAIL PROTECTED])
On Tue, 2006-01-24 at 09:28 -0800, sys read wrote:
right, but how do I pass the rdnis to ccm?
On 1/24/06, Greg Oliver [EMAIL PROTECTED] wrote:
You can have asterisk dial your Unity vmail pilot on busy or
unavailable, and have CCM use the last re
directed number on the trunk to
determine the called extension, or pass the $RDNIS value and
digit
add/strip from * to CCM.
We use * in the exact opposite fashion, but should suffice
either
direction.
On Tue, 2006-01-24 at 07:56 -0800, sys read wrote:
I have my eyes on the Linksys/Sipura 941, ( SIP ), but the
core
problem is that you can't use SIP phones with CCM. I have a
SIP trunk
between asterisk and ccm. I can route calls back and forth,
I just
can't get the call to send to vm if no answer on the
asterisk side.
On 1/24/06, kevin ling [EMAIL PROTECTED] wrote:
Hi,
Maybe buy 7912 phone and register to CCM is another
choice. or
integrated CCM with asterisk voicemail system.
__
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of
sys read
Sent: Tuesday, January 24, 2006 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Asterisk SIP phones to
Cisco
Unity via CCM 4.0SIP Trunk
Hi guys,
I want to leave messages on our unity box. I have
already
converted a couple 7940s to SIP, but I can't give
them out to
our users because I don't want to have to deal with
two
voicemail systems.
we have licenses for all our users on unity as
is.we're
about to buy a bunch more 7940s, but I don't want to
cause
they're expensive. I'd rather buy a cheaper SIP
phone and
have it rollover to the unity vm.
On 1/23/06, Gary Richardson
[EMAIL PROTECTED]
wrote:
You can run a SIP image on a 7940.
[EMAIL PROTECTED] has
pretty good
support for it. Check the voip-info.org wiki
for
instructions on
switching the firmware.
Hopefully that will take a step out of the
plan -- you
could
completely ditch your Cisco system :)
On 1/23/06, sys read [EMAIL PROTECTED]
wrote:
Hi,
I've got a CCM ( Cisco Call Manager ),
with a Cisco
Unity VM server and
about 45 SCCP phones on the ccm, and 200
users on
unity. we want to
migrate all users to IP Phones to ditch
our ancient
phone system. I would
love to get Linksys-Sipura SPA-941s for
the 150
users not on IP phones yet
and run sip to an asterisk server, but
have their
voicemail on Unity.
these phones are $150 each, the
alternative is cisco
7940s ( around $250 )
running SCCP through the CCM. at the
quantities I'm
talking about, $100 is
significant.
Does anyone have any idea how to get this
done?
I've tried this:
exten = 123,1,Dial(SIP/sipphone,20)
exten = 123,2,Dial(SIP/ccm/3040)
where 3040 is our VM pilot for ccm. but
all it does
is take us to the main
Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk
Post your relevant config section and your CCM trunk settings as well as route patter settings. On Tue, 2006-01-24 at 12:16 -0800, sys read wrote: Greg, appending the number just gives me a fast busy. Mike, a) is out because the cheaper cisco sccp phones don't have two way speaker phone b) is what I have, and are trying to get to work. see my previous email about the sip trunk. I don't know what to do to make unity go into the greeting for the user who was called. c) I'd like to, but I just switched voice mail for all my users, and I don't want to endure the nightmare of switching again. that's my long term goal. On 1/24/06, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Options would appear to be: a) use cheaper SCCP phones like 7905/7912 and stay with CCM b) put an asterisk box up and configure a SIP trunk between CCM and Asterisk - I have done this and it works although there used to be a bug with the CCM box not tearing down the RTP at the end of the call - it appeared to rely on receiving an ICMP port not reachable from the other end - this could probably be fixed with the appropriate rtptimeouts ? You would add new users on Asterisk using SIP phones and have a mixed system. c) ditch the CCM and go 100% Asterisk You might consider (b) in the short/medium with a road-map towards (c) Mike - Original Message - From: sys read To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 24, 2006 3:56 PM Subject: Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk I have my eyes on the Linksys/Sipura 941, ( SIP ), but the core problem is that you can't use SIP phones with CCM. I have a SIP trunk between asterisk and ccm. I can route calls back and forth, I just can't get the call to send to vm if no answer on the asterisk side. On 1/24/06, kevin ling [EMAIL PROTECTED] wrote: Hi, Maybe buy 7912 phone and register to CCM is another choice. or integrated CCM with asterisk voicemail system. __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sys read Sent: Tuesday, January 24, 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk Hi guys, I want to leave messages on our unity box. I have already converted a couple 7940s to SIP, but I can't give them out to our users because I don't want to have to deal with two voicemail systems. we have licenses for all our users on unity as is.we're about to buy a bunch more 7940s, but I don't want to cause they're expensive. I'd rather buy a cheaper SIP phone and have it rollover to the unity vm. On 1/23/06, Gary Richardson [EMAIL PROTECTED] wrote: You can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty good support for it. Check the voip-info.org wiki for instructions on switching the firmware. Hopefully that will take a step out of the plan -- you could completely ditch your Cisco system :) On 1/23/06, sys read [EMAIL PROTECTED] wrote:
Re: [Asterisk-Users] T3 Mux and Asterisk Question
I am unsure of * capabilities on NFAS (we do not use PCs to terminate any PRIs), but it allows bonding of desparate PRIs to use a single d-channel. ie, you can have 1 d-channel (optional backups) for the entire DS3. Not sure if * can communicate across cards like that in the same bus though. On Sun, 2006-01-22 at 23:42 -0500, Greg Boehnlein wrote: On Sun, 22 Jan 2006, Steve Totaro wrote: I have a T3 coming from my carrier. From there I want to use an Adtran mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad T1/PRI equipped servers. Everything seems very straight forward with the exception of the D channels for the T1/PRI. I am not very familiar with large circuits such as T3s. I know that I can use one D channel per set of quad port on each server. So if each server has a quad port card, I can use one channel as the D channel for all four spans. That gives me seven D channels in my setup. Does anyone know how the Mux handles these D channels onto the T3? My guess is the Mux is simple going to send all of the channels onto the T3 without modifying anything. That's correct. The T1 spans on the DS3 are completely independent of the clocking on the DS3. The D-channel and timing is something that will be handled by your upstream Telco and the switch that you'll be connecting to. Or, your own box.. ;) What I would really like to do is have one D channel coming in on the T3 and have it split between each of the T1/PRI or even better one D channel per quad (I know Asterisk can do that). Is it possible? No. If the Adtran mx2800 cannot do it, is there anther product that can. I have looked at the RAD Optimux T3 product but have had great experience with Adtran products. The price is the same but the Adtran allows for two controller cards so it seems to have more built in redundancy. Any tips would be appreciated. Adtran's MX-2800 is our choice for Muxes. They are solid, reliable and work well. Adtran's technical support is amazing. When you purchase an MX-2800, you are immediately given access to the Adtran Carrier support group, which doesn't even blink about sending out an advance replacement unit overnight if you ask. ___ --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] setting Cisco 7940 to factory default
Is it set for DHCP - or static? If dhcp, just put option 150 in the scope for a tftpserver on your network. The password can be changed in the config file it asks for. On Tue, 2006-01-17 at 13:38 -0800, Hoss Bazargani wrote: Hi Cory thanks, I bought many 7940 from e-bay for our internal use. The last two I bought, the default Cisco password was changed so I can not use them. One is configured with MGCP and the other one is a SIP. I am configuring the phones in SIP format. so I am not familiar with CallManger and new to VoIP. Just started adminstrating and adding phones. If you have any suggestions or comments please let me know. I am willing to spend $10 per phone but more than that I am forced to sell them back in e-bay and buy a new/refurbished ones. regards Hoss - Original Message - From: Cory Andrews To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 16, 2006 7:14 PM Subject: Re: [Asterisk-Users] setting Cisco 7940 to factory default Hoss - You need to use ethereal / packet sniffer to see what IP address the phone is looking for when it boots up. It should look for a TFTP server, in a CallManager environment the phones check for updates. Once you have the IP address it is looking for, you need to force feed it a new config file. I have posted a step by step procedure for this in the past, I'm not sure if there is an archive somewhere for the Asterisk-Users list, if there is, you should be able to search it and find past posts from me regarding resetting MGCP phones. If you need more help, shoot me an email off list, and I will track down the procedure documentation. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Hoss Bazargani To: asterisk-users@lists.digium.com Sent: Monday, January 16, 2006 6:41 PM Subject: [Asterisk-Users] setting Cisco 7940 to factory default Hi I have two quesitons: 1. I have two Cisco 7940 Phones that I would like to reconfigure. Unfortunatly the default factory password cisco has been changed to something else. Can anyone tell me/guide me as to how to factory default my phone so that I can make configuration changes. I am using SIP but one of the phones has been recofigured to MGCP. 2. I am already using Asterisk with my 7940, what do I need to configure so that I can use Cisco 7905 as well/ Hoss Bazargani 760-305-7000 ext. 405 760-732-3587 Fax [EMAIL PROTECTED] www.ip3.com __ ___ --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 ___ --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
[Asterisk-Users] Asterisk RTP Bridging
I know from everything in the past I have read, that Asterisk natively bridges calls between endpoints. We use * for only ACD and VMail purposes at this point, and I was wondering if there was any way to get a call from: PSTN-MGCP(cisco)-CCM-*(ACD)-Dial(SIP/)-CCM-(CCM phone) to not be bridged after the CCM connected phone answers. TIA for any help. -Greg ___ --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] Cisco dtmf
I use: # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 dtmf_inband: 1 dtmf_outofband: never dtmf_avt_payload: 101 and it works well for me. Sometimes going through a callmanager I have to set outofband to avt to get dialtone sent though. On Tue, 2005-12-27 at 16:05 +0100, Tomislav Parcina wrote: I'm trying to set up call transfer and automon options. They work fine with ZAP lines (analog telephone) and with Grandstream Budgetone 102. I have problem with Cisco 7905 and 7940. I think that problem is with dtmf signalization. This is my configuration in 7940 dtmf_inband: 1 dtmf_outofband: none dtmf_db_level: 3 And 7905 AudioMode:0x Is my configuration wrong or it doesn't work with Cisco phones? Thank you for your time! ___ --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] Cisco Call Manager and H323 trunk correction (MTP)
If using CCM = 4.0, using SIP trunks will alleviate a lot of headaches. On Tue, 2005-11-15 at 16:33 -0800, Dan Austin wrote: I posted a couple weeks back about our experiences with H323 trunks on CCM. As of version 4.0, the Cisco documents state that a 3rd party H323 gateway requires a Media Termination Point.. At the time I said that I have Asterisk working with the ooH323c version of chan_h323 with out an MTP. I just found that another engineer had been twiddling with the CCM config, and we were using a MTP. I retested chan_h323 without the MTP, and indeed per the Cisco docs, when a phone connected to CCM puts a call placed through chan_h323 on hold, the call is disconnected. This IS NOT a bug with asterisk or the chan_h323, but a known Cisco quirk. Cisco's own H323 gateways are capable of dynamically creating/connecting to a MTP. Which permits calls to/through them to allow rtp re-invites and still preserve a call during media transitions. I thought I should post this for the archives in case anyone searching for details about connecting CCM to Asterisk found my earlier misinformation. Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 7940 paperweight
Do you have a XmlDrfault.cnf.xml file on your tftp server? On Fri, 2005-11-11 at 16:02 -0700, Kris Edwards wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Still looking for any advice with this. I had given up with the upgrade process (to SIP.. tftp won't send the files for some reason) but I can't even get this to work with sccp. It doesn't seep to ever finish booting. My understanding is that after the hunt is exhausted through tftp, the phone will boot it's current image, but this isn't the case for me. The display shows Configuring IP Requesting Configuration Opening 192.168.1.104 (tftp server i assigned) Defaulting CM to TFTP Server infinite loop Here is my phone info and below that is a tcpdump. If you have any ideas, please let me know. If this phone is bricked, I need to get my money back before it's too late. MAC Address 00XXBD4D Host Name SEP00XXD4D Phone DN App Load ID P00306000400 Boot Load ID PC0303010100 Version 6.0(4.0) Expansion Module 1 Expansion Module 2 Hardware Revision 4.3 Serial Number INMXXT Model Number CP-7940G Codec ADLCodec Amps 5V Amp C3PO Revision 2 Message Waiting NO excerpt from network settings... CallManager 1 CiscoCM1 CallManager 2 TFTP192.168.1.104 DHCP Enabled Yes DHCP Address Released No Alternate TFTPYes Erase Configuration NO Forwarding Delay NO GARP Enabled Yes Voice VLAN EnabledYes Auto Line Select Enabled No Video Capability Enabled No DSCP For Call Control default DSCP For Configurationdefault DSCP For Services default Device Security Mode Non Secure Web Access EnabledYes Tx Excessive Collisions 0 Tx Frames 232 Tx Broadcasts 28 Tx Multicasts 13 Tx Collisions 0 Tx Deferred Abort 0 Rx Overruns 0 Rx Long/CRC 0 Rx Frames 54 Debug display: 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 Socket Task 616 of 1200 Phone Task916 of 4000 RTP Task 104 of 1200 TLS Task 104 of 6000 Config Task 1592 of 6000 Display Task 472 of 1300 CAST Task 144 of 1600 Sidecar Task 348 of 1500 Audit Task436 of 1600 Undefined Mode0 of 64 SVC Mode 12 of 64 IRQ Mode 28 of 128 FIQ Mode 0 of 64 DomainsnmpUDPDomain Remote Address/0 Local Address /0 Sender Joins 0 Receiver Joins0 Byes 0 Start Time0 Row StatusNot Ready Name SEP00XXBD4D Sender Packets0 Sender Octets 0 Sender Tool None Sender Reports1 Sender Report Time0 Sender Start Time 0 Rcvr Lost Packets 0 Rcvr Jitter 0,0 Receiver Tool None Rcvr Reports 1 Rcvr Report Time 0 Rcvr Packets 0 Rcvr Octets 0 Rcvr Start Time 0 Here is a tcpdump (mac changed): 15:48:31.501856 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:35.501998 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:39.502162 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:43.502293 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:47.504194 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:51.502542 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:55.502685 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:59.502815 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:03.502961 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:07.544093 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:08.033496 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:09.033501 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:11.033569 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:19.864393 CDPv2, ttl: 180s, Device-ID 'SEP00XXBD4D'[|cdp] 15:49:22.922753 IP
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. -Greg On Wed, 2005-11-09 at 16:10 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g722ulaw On asterisk (if this is a right example of pass-thru utilization), I download the codec from http://kvin.lv/pub/Linux/Asterisk/freebsd/ (my processor is a Sempron 2.2, then I download codec_g729-gcc-athlon-sse.so and codec_g729-gcc-debug.so files) and put it in my codec directory /usr/local/lib/asterisk/modules/. I remove the dummy codec first, then on sip.conf: disallow=all allow=g729 allow=alaw allow=ulaw The ISP sip services have support of g729. When I try to make a call from cisco phone to ISP, I see something on CME that seems codec g729 doesn't work: barahir#sh voice call summary PORT CODECVAD VTSP STATEVPM STATE == === == 2/0.1 - - - 2/0.2 - - - 2/1.1 - - - 2/1.2 - - - 50/0/1 .1 g711alaw n S_CONNECT EFXS_CONNECT 50/0/1 .2 - - - EFXS_ONHOOK 50/0/2 .1 - - - EFXS_INIT 50/0/2 .2 - - - EFXS_INIT 50/0/3 .1 - - - EFXS_ONHOOK 50/0/4 .1 - - - EFXS_ONHOOK 50/0/4 .2 - - - EFXS_ONHOOK Where is my mistake? Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDchFRMakHrsrHP9wRAoElAKDxrAxtMOOyRLO6kWaG/hvLVwAj8QCfW/TO LkuPpXb7DVpjUkoi6uV1PNU= =qwXR -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] problem with g729 and CME-Asterisk
Post up your dial-peer 500 config as well. It is doing codec 0x2 (g.711Alaw) from the get go. Also post relevant config for the phone from asterisk and dialplan entry used. -Greg On Wed, 2005-11-09 at 17:08 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me? http://www.nesys.it/snap/debug_voice_ccapi.txt thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDch8XMakHrsrHP9wRAkO2AJ9W15cGdtnWF+oWl0Yd/ai7HTHs+wCg1oUD X8BxszRaAVFpPkQzd1w5jEg= =Jsnv -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsoreby Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] problem with g729 and CME-Asterisk
Just put codec g729(whatever version you need) in your dialpeer. I do not see what the voice-class codec 1 is without that section. -Greg On Wed, 2005-11-09 at 17:17 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDciEJMakHrsrHP9wRArwvAJ9/lz+D1xVL8WnU3dyNLfpkh62nJwCgm8DD /9HE2UKACZ/OOJkZpC8c6Ss= =+5Iw -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Zaptel T1 Timing Source
Circuit timing is only to let the hardware know how to keep in sync with framing and what it is supposed to be. T1 timing will always be the same, so syncing your card to any of them will be fine. Syncing to 2 - 1 as backup would be best, etc.. Timing has nothing to do with the remote end - it remains local. If you have your own stratum source, you could use it instead of line. -Greg On Wed, 2005-11-09 at 13:06 -0500, Waldo Rubinstein wrote: These are REAL Telco T1s and not connected to a PBX. Am I to assume that even if they are different providers the timing should be the same? That doesn't make a lot of sense to me. Thanks, Waldo On Nov 9, 2005, at 12:34 PM, Bart Fisher wrote: My understanding there should only be one timing source per TE410. You should use a REAL Telco T1 for a timing source. - Otherwise, do not choose any if for example all PBX T1's installed. The settings is only a priority level for asterisk to obtain the source. Example: 1 = use this source first choice, 2 = use this source if source 1 is down, and so on.. Bart - Original Message - From: Waldo Rubinstein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 9:12 AM Subject: [Asterisk-Users] Zaptel T1 Timing Source Hi guys, I have a question about the timing source parameter in zaptel.conf. I have 4 T1s coming into a TE410P. One T1 is with one carrier, who provides timing signal. The other 3 T1s are from a different carrier, all sharing the same timing signal. Based on this, I have in /etc/zaptel.conf something like: span=1,1,0,esf,b8zs em=1-24 span=2,1,0,esf,b8zs em=25-48 span=3,2,0,esf,b8zs em=49-72 span=4,2,0,esf,b8zs em=73-96 What I have done is set the timing source of the first T1 to be the primary source for itself. For the other three T1s, I set the second T1 to be the primary source for the group of 3 and the other two as secondary sources. Is this correct? The reason I ask is because every so often I hear people complaint about call drops. It doesn't happen to everyone, so I don't know if it has anything to do with time source selection and synchronization issues that may be affecting individual channels. After a report of a call drop, I check dmesg and I don't really see any errors. Sometimes I just see ... disable echo cancel... messages on specific channels, but that shouldn't be a reason to drop a call. Am I right? Any ideas? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Cisco 7960 Password Recovery
It is set by your SIPMAC.cnf file. phone_password: password ; Telnet/Console Password On Tue, 2005-11-08 at 08:51 -0500, Polycom User wrote: i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work. Thanks in advance ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: Cisco 7970
No - only 323 until CCM 5.0 On Tue, 2005-11-08 at 21:42 -0500, Jonathan k. Creasy wrote: I thought there was a sip image for that phone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Reynolds Sent: Tuesday, November 08, 2005 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Cisco 7970 Jeremiah, You say you have your 7970 working great with * ... The 7970 only supports SCCP, so are you using the chan_skinny modules that come with *, or are you using the chan_sccp modules? Thanks for any response. JR On 11/8/05, Jeremiah Millay [EMAIL PROTECTED] wrote: I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, [EMAIL PROTECTED] wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Cisco 7970
The 7970 when reset to factory will delete the firmware load leaving
just the bootloader.
1. Hold down the # key
2. Power it on
3. Keep holding the power key until the line keys blink orange down the
tree
4. Have the firmware files on your tftpserver when it boots
5. Put the load into the config file like so:
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
It will retrieve the firmware and boot.
-Greg
On Mon, 2005-11-07 at 09:50 -0500, Dan Levine wrote:
Hello
I have a Cisco 7970 phone that when I was trying to reset it to factory
defaults it rebooted and now is stuck in a constant loop of the lights
flashing by going down the line pool one light at a time in a constant
rotation.
I have the firmware for the phone, but have no idea on how to load or it
how to get this phone functioning again.
I would definitely be willing to pay someone to help me get this thing
back online, if someone can contact me either here or offlist to get
this resolved I would appreciate it tremendously.
Thanks
Dan
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
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Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
Forgot to mention - it is 7.0.2-0S firmware
On Fri, 2005-11-04 at 11:35 -0600, Greg Oliver wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
I have not loaded the 7.1 firmware - it must have just recently been
released - it was not on their site last week, so I could not tell you,
but will load it up and play with it over the weekend.
I am running * CVS-HEAD with cnah_sccp-20050922 - my MWI is working as
well as service URL. Does the phone have the correct entry for it when
it boots up under the Settings-3-2 screen?
Are there any relevant status messages on the phone in regards to it?
On Fri, 2005-11-04 at 20:45 +0100, René Enskat [Teamware GmbH] wrote:
Hmm i tried your config but the service url ist still not working.
i have the 7.1 images on the phone.
and the message waiting icon is nothing there too but i have a new message on
the server
On Fri, 04 Nov 2005 11:35:32 -0600
Greg Oliver [EMAIL PROTECTED] wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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Re: [Asterisk-Users] Cisco phone firmware
You probably do not need firmware. I have tried several versions on 70s, 60s, 12s, 05s and 20s (not 02s) with success. If they are not even looking for TFTP, then from the phone, hit Settings-2**#, and erase. Make sure your DHCP server is kicking out option 150 right (the correct TFTP server) - if it is Linux, you will need to create several symbolic links due to the differing upper/lower case requests from differing phones and firmware versions, but the XmlDefault.cnf.xml should get them going fine. Do a tcpdump from the server to make sure they are actually requesting the files. Setting-2 should list the correct DHCP / TFTP servers. On Fri, 2005-11-04 at 15:51 -0600, Ryan Amos wrote: I understand that I must pay for a support license to download Cisco firmware, so I’m not trying to pirate it. I simply want to know what I need to buy in order to get firmware files for my phones. Does anyone have any helpful links they can give? What does this license cost? Specifially, I need SCCP images for the 7902 and 7940… Does anyone have any experience with setting this up and can either give me a hand or point me in the appropriate direction? Most of the entries on voip-info.org were pretty light on details, just postings of different config files. Also, if I’m just doing skinny (no need for SIP, and the 7902s don’t support it anyway) do I even need firmware files? It seems like I do, but I can’t even seem to get my phones to try to tftp in and download anything. Also, for anyone who has used either of these phones, how well do they work with asterisk? The chan_sccp2 drivers say they “mostly work” but I want to know what doesn’t work to see if I care. Any help would be appreciated, thanks. -- Ryan Amosing System Administrator, FineTooth http://www.finetooth.com/ 512-637-3530 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MTP required for CCM integration ?
You will probably also need to change the media exchange timers in CCM if you are going to use it as a PRI gateway - otherwise asterisk - 323 - CCM - PSTN calls will get dropped after 4 secs of ringing. On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote: Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP. I will continue my tests, and maybe give a try to the patch you mentionned. However, this will probably be too cutting edge for the project ;-) I have a few questions, though: - You mention that Cisco indicates that any H323 trunk with advanced features needs an MTP. Can you point me to the place where you found this ? Because as far as I can tell, this is not true for a trunk to a Cisco gateway. - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, I should have better luck with the Sourceforge version... - From your experience, do you feel that a clean CCM-* integration is possible ? I am currently interested in simple feature (MoH, transfers, maybe Call Park). A friend of mine is working on the voicemail (unity) replacement/integration. Thanks again for you quick support, and sorry for my late answer ! BR, - Patrick - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: vendredi, 21. octobre 2005 18:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? Is it required to use an MTP on the Cisco callmanager, when integrating with asterisk (using h323) ? As of CCM 4.X, Cisco indicates that any H.323 trunk that will support MoH/Transfer/etc need MTP resources. Annoying. I am working on a project where the goal is to interconnect Cisco Callmanager (version 4) clouds together, using either SIP or IAX between multiple * servers. Basic setup will be: PHONE - sccp - CCM (V4) - h323 - ASTERISK - iax - ASTERISK - h323 -CCM - sccp - PHONE I am working on the first half of it, which is: 7920 --- SCCP --- CALLMANAGER (V4) --- chan_oh323 --- ASTERISK 1.0.9 I want to avoid the use of a gatekeeper. In that configuration, I am trying to get call transfer working. The phone can call the DEMO app on asterisk, but then I cannot transfer the call to another Cisco phone (on the same callmanager). I have some PCAP traces if required. Basically, the 2nd phone rings, but there is no audio channel. After about 10 seconds, I see that that chan_oh323 hangs up the call. Sure will drop the call. MTP does solve this. The idea was to avoid RTP streams through the call manager. Good plan, and one that I would consider a must for scalability and quality. Now, if I define a Media Termination Point (MTP) on the Callmanager, things work much better. I have also tried the new ooh323 with 1.2.0-beta1, but I couldn't get audio at all. Odd, I am using ooh323c. I have a special test release, but the fixes for our CCM4 enviroment were added to CVS. Are you using ooh323c from Asterisk-Addons or a download from Open Systems? I have read a lot about people having success with integratin CCM and*, but without any details, especially around MTP configuration. Any help would be greatly appreciated. BR, - Patrick - http://bugs.digium.com/view.php?id=5374 has a patch that allows * to send RTP packets when it is not receiving them. I wasn't expecting this result, but applying this patch resolved the disconnect when a SCCP phone put a call on hold and allows transfers. The bug/patch got quite a bit of early attention, but seems to have languished. Try it out and provide feedback. Maybe enough success reports will help get it rolling again. Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 IVR and Cisco Call Manager
With asterisk and call manager hooked up via the sip trunk, the calls from ccm and asterisk can call each other. I have 2 problems. 1. Is it possible to route all calls via the call manager and not via asterisk when I dial any number? Yes 1. This is divided into 2 problems a. I know when u dial into call manager and press a number, you can forward to asterisk but can the asterisk ivr service process the request and route back to call manager to make the call via the call manager? Somehow the problem 1 and 2 are related. Yes a. Is this doable with the sip trunk? Yes Contact me offline with any questions. Regards, Dinesh. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/Cisco Call Manager 3.3
I would have to agree - your easiest route is to upgrade to CCM 4.0+ with SIP trunk support.. On Fri, 2005-10-14 at 16:55 -0500, Paul Davidson wrote: Message: 13 Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk/Cisco Call Manager 3.3 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco Call Manager 3.3 does not support sip trunking. Is there anyway this can be done. Please shed some light on this topic. Thanks. Goran Goran- Speaking from experience, you have a tough road ahead of you. The only way to accomplish this is via h.323 trunks under Cisco and Asterisk. There are a few known good configurations- I can really only speak to one, as it eventually worked for me- but others may have different and perfectly reasonable advice. First, some prerequisites: 1. Asterisk 1.2 or CVS HEAD. Do NOT try this with any of the 1.0X series- you will be able to call from CCM to Asterisk, but not from Asterisk to CCM. 2. An H323 Gatekeeper. GnuGK works, but does occasionally bonk out. CCM will send RRQ requests to the gatekeeper at a rate of 10x per second, and eventually, GnuGK loses it. An IOS gatekeeper seems to be much better. 3. chan_h323 set up and running properly. There's whole readme files on the prerequisites for this- read them, follow the directions closely- and call on JerJer *LAST* if you value your life. 4. A Gatekeeper controlled Trunk on CCM. The tricky bits here are the significant digits, and the technology prefix. CCM does *NOT* register the tech prefix or it's extensions with the gatekeeper- so you'll have to config the gatekeeper to know where to send the call, and you'll have to configure your CCM dialplan to act accordingly. Set this up slowly. Get a working Asterisk box that's able to handle softphones or hardphones as an island PBX, then configure the H323 trunk- you'll save some frustration of trying to configure both simultaneously. Find me on the IRC channel if you need specific questions answered- or email me directly. I can optionally configure it for you for a fee- I'm based in the US, and judging from your accent, I'd say you aren't- I can do this remotely if needed. I won't charge you for questions answered. :) -pbd ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 7960g 2nd ethernet port cycles on/off
What type of switch/hub is it connected to? On Thu, 2005-10-06 at 15:40 -0700, Tom Tune wrote: I saw a thread from 2003 that addressed this problem but they didn't post a fix: When I plug my PC into the 2nd ethernet jack on my Cisco 7960g it loses connection on and off for ~30 seconds at a time. I tried playing with setting the speed from auto to 100/full, half, etc. to no avail. Any help on this would be appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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, cisco 3640's and DIDs...
Add direct-inward-dial to your dial peer and it should work fine. -Greg On Mon, 2005-10-03 at 15:48 -0700, Tim Pozar wrote: I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day cisco is all lower case :-)). My config looks something like this on the cisco... - voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 description RCN PRI at SF7 ! interface FastEthernet1/0 no ip address duplex auto speed auto ! interface Serial3/0:23 no ip address dialer-group 1 isdn switch-type primary-5ess isdn incoming-voice voice no cdp enable ! voice-port 3/0:23 connection plar 1000 ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.2.3.4:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2 pots destination-pattern 9T port 3/0:23 ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:1.2.3.5 ! - But of course with that what get's set as the DID number is 1000. I need to find out how to get the DID number passed to asterisk. Any thoughts from folks out there? Thanks... Tim PS... Here is a show ver from the 3640... vr01-200p-sfoshow ver Cisco Internetwork Operating System Software IOS (tm) 3600 Software (C3640-IS-M), Version 12.3(16), RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2005 by cisco Systems, Inc. Compiled Tue 23-Aug-05 20:03 by ssearch Image text-base: 0x60008B00, data-base: 0x61BFA000 ROM: System Bootstrap, Version 11.1(19)AA, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1) ROM: 3600 Software (C3640-IS-M), Version 12.3(16), RELEASE SOFTWARE (fc4) vr01-200p-sfo uptime is 3 weeks, 3 days, 22 hours, 22 minutes System returned to ROM by reload at 00:24:09 UTC Fri Sep 9 2005 System restarted at 00:25:48 UTC Fri Sep 9 2005 System image file is flash:flash:c3640-is-mz.123-16.bin cisco 3640 (R4700) processor (revision 0x00) with 124928K/6144K bytes of memory. Processor board ID 10311643 R4700 CPU at 100MHz, Implementation 33, Rev 1.0 Bridging software. X.25 software, Version 3.0.0. SuperLAT software (copyright 1990 by Meridian Technology Corp). Primary Rate ISDN software, Version 1.1. 2 FastEthernet/IEEE 802.3 interface(s) 24 Serial network interface(s) 1 Channelized T1/PRI port(s) DRAM configuration is 64 bits wide with parity disabled. 125K bytes of non-volatile configuration memory. 32768K bytes of processor board System flash (Read/Write) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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, cisco 3640's and DIDs...
Hm, I would have to disagree. We use MGCP dial-peers and use it on PRIs with 3725s and 2851s currently. On Tue, 2005-10-04 at 12:12 -0700, Tim Pozar wrote: Greg Oliver wrote: Add direct-inward-dial to your dial peer and it should work fine. That command is only supported for POTS interfaces. :-( Not PRIs (aka ISDN in cisco parlance). Tim ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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, cisco 3640's and DIDs...
Glad to hear it! On Tue, 2005-10-04 at 17:53 -0700, Tim Pozar wrote: Greg Oliver wrote: Hm, I would have to disagree. We use MGCP dial-peers and use it on PRIs with 3725s and 2851s currently. Our config was fubar'ed. We were using dial-peer isdn instead of pots. direct-inward-dial does not work with isdn. We were succussful with something like: -- isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 description RCN PRI ! interface Serial3/0:23 no ip address isdn switch-type primary-5ess isdn incoming-voice voice no cdp enable ! voice-port 3/0:23 ! dial-peer cor custom ! dial-peer voice 4 voip ! We are matching the four digits that RCN is handing us going from ! 2100 to 2199. This needs to happen else the cisco will tend to route ! calls back out the PRI. :-) destination-pattern 21.. session protocol sipv2 session target ipv4:1.2.3.4:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 200 pots destination-pattern .T direct-inward-dial port 3/0:23 forward-digits all ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:1.2.3.4 ! -- With this we have DIDs working. Thanks to [EMAIL PROTECTED] for the sample configs. Tim ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] strange wave like noise on sip handset
We have all Cisco - and they are pricey, but work great otherwise. Both with chsn_sccp and SIP. 05 - 70s and a few 20s -Greg On Sat, 2005-10-01 at 14:47 +0100, Angus Comber wrote: No it happens on our asterisk and at a customers. Not that noticeable but not crystal clear. Didn't happen on a Snom 190. I have been working my way through IP handsets with these results: Grandstream BT-100 series. OKish for the price but a bit echoy. Grandstream GXP-2000 - OK but if used on hands free a bit echoy. Snom 190. Very clear. However, on a customer site they complained that full volume was still not load enough. But didn't extensively test. Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds at start of call then echo went away. Remote end did not hear any echo. Also wave like hiss as per my message. Next phones to try are a Polycom 300 and a CISCO 7940. I suppose it depends on how demanding customer is. I would hope that I can find a phone with no echo / hiss /other problems. Perhaps I need to think about using channel banks/FXS cards and analog phones! But would prefer IP phones for flexibility etc. Anyone found a perfect IP phone? Angus - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 01, 2005 2:33 PM Subject: Re: [Asterisk-Users] strange wave like noise on sip handset On 9/30/05, Angus Comber [EMAIL PROTECTED] wrote: On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? I heard the same thing from a remote users Polycom 501 - seems it was sitting too close to a fan in a computer. Could it be something similar to that? Just a thought since this happened to me yesterday :) -- Leif Madsen - http://www.leifmadsen.com Astricon 2005, Anaheim, CA, October 12-14 http://www.astricon.net http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] cisco phones problems
Whatever you have the voice vlan set it is what they operate on. You cannot provision that on the phone manually. If they are small switches (35xx, etc), then you need to configure without .1q trunking as those switch imply it automatically. For the larger switches 1.q trunking in the config is required for phones to properly operate on dhcp and the pcs attached to function properly. On 14:51, Fri 30 Sep 05, Edwin Lam wrote: after much struggles. i've found out that if i ping the phone unit from another computer constantly (couple pings every 5-10 sec) the phone will operate fine. once i stopped the pings, the UNREACHABLE message started to pop up and the drop calls problems starts. seems like it's the firmware issue. does anyone uses Cisco SIP 7.3 (or 6.0, i've tried downgraded it at some point) and have similar problems? p.s. another piece of info: the phone units are set to a non default vlan manually since we share the physical lan for both data voice. Hi, I had the same problem with only 1 Cisco 7905 every once in a while. All problems were solved as soon as I reverted the phones to SCCP and started using chan_sccp.so There's no lag anymore between the phones and asterisk. So maybe this is an extra reason to suspect the firmware ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
