Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
I've no experience with that phone model or protocol. But if you run a tftp trace you'll see what files the phone is looking for. Check my old thread on pbxinaflash forums for details. i -- Original Message -- Received: 04:59 AM COT, 06/16/2011 From: bilal ghayyad bilmar...@yahoo.com To: ianworthing...@usa.net, rswago...@gmail.com, s...@open-t.co.uk, cass...@cassius.org, wcse...@selbytech.com, asterisk-users@lists.digium.com Subject: Cisco IP Phones 7942G (skinny): TFTP and required files Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
Ah-ha! Progress at last. (I'd actually tried debug mode before and wondered why I got no output. Any harm in leaving that console = etc enabled?) Console is showing the following. Looks like it doesn't like the format of the REGISTER message??? --- SIP read from UDP:192.168.1.114:5060 --- REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 From: sip:702@192.168.1.41;user=phone To: sip:702@192.168.1.41;user=phone Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:702@192.168.1.114:5060 Content-Length: 0 Expires: 120 - [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 0 [ 33]: REGISTER sip:192.168.1.41 SIP/2.0 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 2 [ 39]: From: sip:702@192.168.1.41;user=phone [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 3 [ 37]: To: sip:702@192.168.1.41;user=phone [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 4 [ 58]: Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 5 [ 18]: CSeq: 101 REGISTER [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 6 [ 18]: User-Agent: CSCO/7 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 7 [ 37]: Contact: sip:702@192.168.1.114:5060 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 8 [ 17]: Content-Length: 0 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 9 [ 12]: Expires: 120 --- (10 headers 0 lines) --- [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7539 find_call: = Looking for Call ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 (Checking From) --From tag --To-tag [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7543 find_call: REGISTER request has no from tag, dropping callid: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 from: sip:702@192.168.1.41;user=phone [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do: Invalid SIP message - rejected , no callid, len 337 ian ... -- Original Message -- Received: 07:31 AM COT, 05/30/2011 From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3 On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: And f/w POS3-07-4-00 That is strange that Asterisk is not sending anything back in response to the register. Have you looked at the Asterisk console or logs to see why it is rejecting the register. You might have to enable debug mode core set debug 5 sip set debug on Also if you want to see debug output on the screen check that the following is uncommented in /etc/asterisk/logger.conf console = notice,warning,error,debug Is it possible for you to try a later firmware version? Although 7.4 looks to be a good version according to others notes. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
Many thanks for that. I tried pedantic=no (adding it directly to the [702] section in sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to have a way to enter that through the gui), but it didn't fix it: same console log. Where might I find a reliable source for f/w 8.12? I'm a bit nervous about that as I read that some people feel 7.5 was the last reliable version, and that once you go to 8.x you can't go back? i -- Original Message -- Received: 03:53 PM COT, 05/30/2011 From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3 On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington ianworthing...@usa.net wrote: Console is showing the following. Looks like it doesn't like the format of the REGISTER message??? --- SIP read from UDP:192.168.1.114:5060 --- REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 From: sip:702@192.168.1.41;user=phone To: sip:702@192.168.1.41;user=phone Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:702@192.168.1.114:5060 Content-Length: 0 Expires: 120 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7539 find_call: = Looking for Call ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 (Checking From) --From tag --To-tag [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7543 find_call: REGISTER request has no from tag, dropping callid: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 from: sip:702@192.168.1.41;user=phone [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do: Invalid SIP message - rejected , no callid, len 337 The log states find_call: REGISTER request has no from tag, dropping callid. If you look at the From: line, it should end with ;tag=SOMEVALUE. Looking at sip.conf you could set pedantic=no and the phone should register. The best solution would be to upgrade the phone firmware. I know 8.12 works. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
Sincere thanks Ryan: all is working at long last. I risked the f/w upgrade path in the end rather then something which will be blown away at the next upgrade and leave me scratch me noggin in confusion. Couldn't have done it without your insight. Thanks again. i -- Original Message -- Received: 05:11 PM COT, 05/30/2011 From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3 On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: Many thanks for that. I tried pedantic=no (adding it directly to the [702] section in sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to have a way to enter that through the gui), but it didn't fix it: same console log. The setting is a global setting. With FreePBX you want to add pedantic=no to /etc/asterisk/sip_general_custom.conf You can verify from the Asterisk console with sip show settings. You should see Pedantic SIP support: No under Global Signalling Settings Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
\000\000\000\000\000\000\000\000\000\000\000\000 ... Tried changing, as per your suggestion, to nat=yes and your given settings in both SIPDefault.cnf *and* SIPnncnf without change. ian -- Original Message -- Received: 09:03 PM COT, 05/28/2011 From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3 On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: I too had heard that 1833 did NOT have the 184 problem, which makes me suspicious that it's not that. I don't think its a NAT problem. Neither a sip trace not tcpdump show any response at all to the incoming REGISTER. The phone is on the local lan. I have nat=no and nat_enable: 0 You are running tcpdump on the Asterisk server? Are you capturing all traffic or only certain ports? What firmware are you running on the phone? I am using PS03-8-12-00. It wouldn't hurt to try with nat enabled, see below. I setup all my phones this way as it saves having to reconfigure when users take them home. sip.conf nat=yes SIPDefault.cnf nat_enable: 1 nat_address: nat_received_processing: 1 Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
And f/w POS3-07-4-00 i -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco registration problem with 1.8.3.3
I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER packets sent from phone but no response from Asterisk Is there any way this regressed code could be picked up in a 1833 build or have I got another problem? i -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco registration problem with 1.8.3.3
I too had heard that 1833 did NOT have the 184 problem, which makes me suspicious that it's not that. I don't think its a NAT problem. Neither a sip trace not tcpdump show any response at all to the incoming REGISTER. The phone is on the local lan. I have nat=no and nat_enable: 0 i -- Original Message -- Received: 03:45 PM COT, 05/28/2011 From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3 On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington ianworthing...@usa.net wrote: I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER packets sent from phone but no response from Asterisk Is there any way this regressed code could be picked up in a 1833 build or have I got another problem? I'm able to register a 7940 against Asterisk 1.8.4.1. You might try out that version as it has the fix for registering Cisco phones. However I thought the bug was introduced in 1.8.4 and not 1.8.3.3. I know in the past when I had issues registering Cisco phones I had to make sure the nat settings matched. If you set nat=yes in the sip.conf you must set nat_enable: 1 in SIPDefault.cnf for the phone. What I noticed was when nat=yes is set in Asterisk it ignores the rport and always sends the reply on the port used for the request. Cisco will ignore this reply and not register. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
