[asterisk-users] Cisco 7970 with skinny on * 1.4.x
Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --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 with skinny on * 1.4.x
A little with skinny debug set to on shows during register: Device SEP00175A872053 is attempting to register Requesting capabilities Buttontemplate requested Adding button: 9, 1 Sending 30006 template to cisco Received SoftKey Template Request Received SoftKeySetReq RECEIVED UNKNOWN MESSAGE TYPE: c Received CapabilitiesRes Adding codec capability '0 (25)' Adding codec capability '4 (4)' Adding codec capability '8 (2)' Adding codec capability '0 (15)' Adding codec capability '0 (16)' Adding codec capability '0 (11)' Adding codec capability '256 (12)' Adding codec capability '256 (12)' Adding codec capability '0 (257)' Device capability set to '268' RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 4a RECEIVED UNKNOWN MESSAGE TYPE: 9 Received Time/Date Request Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :92 It also show this message when going offhook: RECEIVED UNKNOWN MESSAGE TYPE: 49 Setting ringer mode to '1'. skinny_new: tmp-nativeformats=268 fmt=4 Attempting to Clear display on Skinny [EMAIL PROTECTED] Clearing Display Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :85 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :11 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :9a Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :82 Looks to me that chan_skinny doesn't understand many important messages. Any previous 7970G SCCP firmware that might work? cheers rick Richard Klingler schrieb: Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --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-1.4.3
Hello (o; Did I miss somewhere the announcement of 1.4.3? Also don't see anything in the announce mailing list archive...but it is available for download... So do I need to download to find out what has changed? (o; cheers rick ___ --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-addons-1.4 write wrong uniqueid
Evnin' As I didn't find any answer I'll try to rephrase the problem (o; Any idea why the latest asterisk-addons-1.4 write wrong uniqueid into mysql database? Asterisk-1.4.2 creates call record files with the uniqueid prepended: 1175107269-SIP-999-0876c000.wav But into mysql database it writes an uniqueid of: 1175107260.88 but should be: 1175107269 Any idea why the difference? Any why it even writes it in decimal format? cheers rick ___ --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] ARI with * 1.4.2 won't display recordings
Evnin' Now I tracked my problem down why ARI won't display most of the recordings... It write a recording for examples as: 1175031785-SIP-0615000995-0872a000.wav But it writes to the field uniqieid into MySQL database as: 1175031779.16 WHen I overwrite the uniqueid field with the value from the recording file, the recording is playable within ARI: 1175031785 Any idea why the uniqieid and the ID used for creating the recording files are always different? cheers rick ___ --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] ARI with * 1.4.x
Afternoon A little off-topic...but... Does any1 know why recorded call with IAX2 in the filename are not displayed within ARI? LittleJohn's website isn't a helpful place for ARI (o; cheers rick ___ --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] Remote host can't match request NOTIFY to call
Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick ___ --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 with skinny on * 1.4.1
Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --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] Cisco 7970 with skinny on * 1.4.1
I was able to register to * 1.4.1 via skinny...and it showed up on the lines and devices show output.. On the phone, however, no lines were displayed nor could it phone out or receive any calls... Anyone able to share some snippets of their skinny.conf? I just used the examples and modified the MAC line and extension line config...but seems something else is missing... cheers rick Pavel Jezek schrieb: If you have 7970 right configured to point to asterisk server, you should be able to see some skinny debug on console, or look what report skinny show devices I haven't any 7970, so can't help so much, I'm using only 7920 wifi phone with chan_skinny and 1.4trunk, it's usable, basic functionality is working, but don't expect too much, btw, if you have money to buy this highend phone with proprietary signaling, why don't connect to callmanager? asterisk will never support all features available in proprierary system as good as original ;-) PJ Richard Klingler wrote: Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --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] Cisco 7970 with skinny on * 1.4.1
Bill Hackensack schrieb: On 3/21/07, *Richard Klingler* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... chan_sccp is far from dead and it works with 1.4. more fud being spread... Maybe not dead...but hust won't compile on FBSD (o; cheers rick ___ --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] Cisco 7970 with skinny on * 1.4.1
Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the sample configuration are of no help... Has any1 got their 7970 to work with * 1.4.x ? cheers rick ___ --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] 7970G SIP8-0-4 not registering with asterisk
Evnin' Already asked that a while ago (o; Has someone an explanation why a 7970G running SIP firmware 8.0.2 can't correctly register with asterisk 1.2.11? It registers quickly but asterisk marks it right after registration as unreachable: -- Registered SIP '1002' at 62.x.x.x port 5060 expires 3600 -- Got SIP response 400 Bad Request back from 62.x.x.x Sep 28 23:36:40 NOTICE[28373]: chan_sip.c:11595 sip_poke_noanswer: Peer '1002' is now UNREACHABLE! Last qualify: 0 rick ___ --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] Forcing Marker bit, because SSRC has changed
Trying again Has anyone an explanation why this error happens? Only hear my echo and not the other side anymore... and the other side can't hear me... Version asterisk 1.2.9 -- Executing Macro(SIP/1001-9c43, stdexten|1010|SIP/1010) in new stack -- Executing Dial(SIP/1001-9c43, SIP/1010|40|o) in new stack -- Called 1010 -- SIP/1010-8035 is ringing -- SIP/1010-8035 answered SIP/1001-9c43 -- Attempting native bridge of SIP/1001-9c43 and SIP/1010-8035 == Forcing Marker bit, because SSRC has changed == Forcing Marker bit, because SSRC has changed cheers rick ___ --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] No channels available after reloading config
Evnin' Has someone experienced the same with the FreePBX frontend? After changing a SIP extension and pressing the red bar on top in the browser I only see on the CLI: sip*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 62.x.x.x (None) 689e04a844a 00102/0 unkn No Init: OPTIONS 62.x.x.x (None) 26e4a49e765 00102/0 unkn No Init: OPTIONS 217.x.x.x(None) 63eb8a8316b 00102/0 unkn No Init: OPTIONS 83.x.x.x(None) 2568ed13019 00102/0 unkn No Init: OPTIONS 62.x.x.x (None) 72f34828082 00102/0 unkn No Init: OPTIONS 62.x.x.x (None) 78e8f4ab628 00102/0 unkn No Init: OPTIONS ...and no further calls are possible... Only way out is to completely restart asterisk in the shell... thanx in advance rick ___ --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] Forcing Marker bit, because SSRC has changed
Evnin... Googled around for this strange error meesage with no helpful results at all... Does somebody has any idea what this means? Forcing Marker bit, because SSRC has changed At the same time I only get inbound audio but other side can't hear me...sometimes I just hear my echo and nothing from other side... Asterisk version 1.2.9 and both participants with public IP addresses...so no NAT/Firewall involved... cheers rick ___ --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] Problems getting 7970G upgraded to SIP
Hi Jason loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/loadInformation6 1. Stick with the 8.0.2 SIP image as it works best with asterisk... at least for me (o; - Here are TFTP server logs to illustrate that I'm using the correct case'd XmlDefault.cnf.xml file: Sep 10 21:57:55 bubbles tftpd[89195]: jalc7970.sip : read request for SEP00131A4D39F4.cnf.xml: File not found 2. I thought you created your SEP file? And still it can't be found? Sep 10 21:57:55 bubbles tftpd[89197]: jalc7970.sip : read request for //XmlDefault.cnf.xml: success 3. Wondering what messages are coming after that...or is it the point where it starts over again? - All the files from the .cop are 100% unmodified. I just tar -zxvf cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into the tftpd root directory, which is the same place the SEP and XmlDefault file are located. 4. So you have all those: -bash-2.05b$ tar tzvf cmterm-7970_7971-sip.8-0-2SR1.cop 644 Mar 22 23:49 SIP70.8-0-2SR1S.loads 2538161 Mar 22 23:49 apps70.1-1-1-15.sbn 411264 Mar 22 23:49 cnu70.3-1-1-15.sbn 1996 Mar 23 00:06 copstart.sh 2401588 Mar 22 23:49 cvm70sip.8-0-1-18.sbn 483105 Mar 22 23:49 dsp70.1-1-1-15.sbn 465288 Mar 22 23:49 jar70sip.8-0-1-18.sbn 71 Mar 23 00:06 load119.txt 72 Mar 23 00:06 load30006.txt 0 Mar 23 00:06 signed/ 4046848 Mar 23 00:06 signed/cmterm-7970_7971-sip.8-0-2SR1.cop 644 Mar 22 23:49 term70.default.loads 644 Mar 22 23:49 term71.default.loads Anyone have any ideas? 5. Not yet. But might be you need to go with a firmware in between first before going with 8.0.x. cheers rick ___ --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 trunk
hello If I want to use asterisk to hookup to a SIP account I just use the register line in sip.conf with the extension number at the end... But how about if I want to use a SIP trunk from a provider which gives me 10 DID numbers with the same account? thanx in advance rick ___ --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] Register 2 times with same host
And now for something completely different (o; Is there a way out of a problem when registering 2 times with different account with same host? I've setup 2 seperate peers using seperate context in sip.conf...but as soon I change one extension in one context it influences the other as well and vice versa... Worst scenario is when it doesn't find a local extension and tries to call out on other peer (o; And when does a context in extensions.conf exactly pick up the Start extension? Only when a call comes in with no destination set? cheers rick ___ --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] Caller ID display on 7970G
Hello (o; Ist there a way to remove the trailing @domain from the displayed caller id on the Cisco 7970G? No problem dialing a number from the missed call directory with the domain attached...just looks weird (o; cheers rick ___ --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] netmask
Hi Dean Dean Collins schrieb: I don’t know if I’m mistaken or not but I noticed in a iax2 show peers command that it is showing my iax2 connections as netmask 255.255.255.255 /32 are hosts addresses...which is correct. All of my lan traffic is supposed to be running on 255.255.255.0 This doesn't mean that all hosts on the internet need the same subnet as you (o; How would you or asterisk know what netmask is used on a remote host not on the local subnet? chers rick ___ --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 directories and services xml
Tomislav Parčina schrieb: According to this thread http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3 Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco 7970 IP Phone doesn't show phone directory or services. It seams there is the same problem with SIP 8.0.3 firmware. Has anybody find any solution to this? Or all we can do is to wait new SIP firmware (8.0.4 can't register with Asterisk). My 7970G running 8.0.2 SIP firmware works perfectly with the Open XML 79xx directory frontend... Also can can push XML alarm messages to the phone from nagios system. For me all other SIP version won't register with * 1.2.9 (o; - Do you have access to the webserver logs? - can you telnet to your webserver port and look on the console if something is returned? (telnet x.x.x.x 80 and do a manual get) - Can you point your phone to some other URLs mentioned on voip-info.org? cheers rick ___ --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] Re: Cisco 7970 directories and services xml
Tomislav Parčina schrieb: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... My 7970G running 8.0.2 SIP firmware works perfectly with the Open XML 79xx directory frontend... I have never tried Open XML 79xx, although I have hear of him. http://www.asteriskpbx.de/index.php?open79xx Also can can push XML alarm messages to the phone from nagios system. Can you tell me more about this? http://www.voip-info.org/wiki/index.php?page=Cisco+79XX+XML+Push For me all other SIP version won't register with * 1.2.9 (o; Same here. - Do you have access to the webserver logs? Yes I have. I can open http://10.0.0.20/cisco/services/PhoneDirectory.xml from my web browser and in web server log I can see that it has requested it. When I press directory button I don't get any message in webserver log. - can you telnet to your webserver port and look on the console if something is returned? (telnet x.x.x.x 80 and do a manual get) I can't do this. But like I said before, it shouldn't be problem with http server because 7940 phone gets PhoneDirectory.xml Would be good to know what the actual text output is to compare with mine... Discovered that it doesn't like long names in open 79xx xml dir and also no umlauts are allowed... But if you have php installed you can check with something like: ? header(Content-type: text/xml); header(Connection: close); header(Expires: -1); print(CiscoIPPhoneDirectory\n); print(\tTitleDirectory/Title\n); print(\tPromptSelect Directory/Prompt\n); print(\tDirectoryEntry\n); print(\t\tNameSomeone/Name\n); print(\t\tTelephone1000/Telephone\n); print(\t/DirectoryEntry\n); print(/CiscoIPPhoneDirectory\n); ? cheers rick ___ --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] Wrong CallerID passed to SIP phone
Mornin' (o; Now tried with the Dial flag o which should pass the inbound caller-id to the extension but that doesn't set it as well...still keeps the caller-id configured in sip.conf Any way debugging on the CLI which CallerID is passed on an inbound call via the register command in sip.conf? thanx in advance rick Richard Klingler schrieb: Evenin' (o; Following strange problem: 7970G SIP phone - asterisk - SIP provider In sip.conf I register to my SIP provider to receive calls from them...but as soon the numer rings I see as CallerID the configured outbound number from my SIP account and not who is actually calling... So I gotta lots of missed calls from myself (o; I thought I saw somewhere an option to the Dial command somewhere to pass the CallerID...and oddly I don't see any calling phone number on the CLI with verbosity even set to 8 thanx in advance rick ___ --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] Re: Cisco 7970 8.0.4 SIP firmware
Tomislav Parčina schrieb: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Where did you find 8.0.3 SIP image? Cisco website... I didn't noticed 8.0.3 SIP firmware there... Also on their ftp: -rwxrwxr-x1 518 201 8136838 Mar 6 2006 cmterm-7970_7971-sip.8-0-2-0.cop -rwxrwxr-x1 518 201 8136765 Mar 28 22:20 cmterm-7970_7971-sip.8-0-2SR1.cop -rwxrwxr-x1 518 201 4106360 May 17 20:19 cmterm-7970_7971-sip.8-0-3.cop -rw-r--r--1 518 201 4114898 Aug 29 20:20 cmterm-7970_7971-sip.8-0-4SR1.cop Just tried now with the 8.0.2.SR1 image... Keeps on saying registering Have you tried the one on the end - Another SEPmac.xml.cnf example Actual problem was with the Phonelabel string being too long (o; Found out with in the logs... Then I tested all images again and only 8.0.2 works with asterisk... all other say on the display that they are registered which asterisk acknowledges...but in the same moment it marks them as UNREACHABLE and only outbound calls are possible with images other than 8.0.2. So I'm staying with SIP 8.0.2 as it also supports XML push whereas the SCCP images don't support it at all... thanx for helping rick ___ --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] Matra 6501
EHLO (o; Anyone succeeded with hooking up a Matra 6501 PBX to * ? cheers rick ___ --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] Wrong CallerID passed to SIP phone
Evenin' (o; Following strange problem: 7970G SIP phone - asterisk - SIP provider In sip.conf I register to my SIP provider to receive calls from them...but as soon the numer rings I see as CallerID the configured outbound number from my SIP account and not who is actually calling... So I gotta lots of missed calls from myself (o; I thought I saw somewhere an option to the Dial command somewhere to pass the CallerID...and oddly I don't see any calling phone number on the CLI with verbosity even set to 8 thanx in advance rick ___ --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] PBX - VoIP migration
Morning (o; What would give me less headache for integrating a Nortel PBX to VoIP? a) Hook up with a Cisco which handles the SIP stuff and E1 to telco failover? b) Hook it up to an asterisk box instead? If I would go with plan (b)...is there an option I can sort of pipe through the E1 trunk coming from local telco within asterisk to the PBX but also start hooking up SIP accounts to it? cheers rick ___ --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: Cisco 7970 8.0.4 SIP firmware
Tomislav Parčina schrieb: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does the 8.0.3 image has the same flaws as 8.0.4? Where did you find 8.0.3 SIP image? Cisco website... Wasn't even able to register with * at all since most configuration examples from voip-info.org wouldn't work... Do you have any example config for me to try with SIP image on 7970G? On this page you can find conf examples. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP Another SEPmac.xml.cnf example works for sure, that is cnf file that I use. Just tried now with the 8.0.2.SR1 image... Keeps on saying registering and what's even worse... ...you can't change any SIP setting, even when unlocked... just keeps those settings once loaded from the SEP file and ignores future updates on this file... thanx in advance rick ___ --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 8.0.4 SIP firmware
Does the 8.0.3 image has the same flaws as 8.0.4? Wasn't even able to register with * at all since most configuration examples from voip-info.org wouldn't work... Do you have any example config for me to try with SIP image on 7970G? Only tried 8.0.3 on my 7970G and had to switch to SCCP image...which is now 8.0.4 cheers rick Aaron Daniel schrieb: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
