Re: [asterisk-users] OT: Quad-band cellphones with wifi stable sip support
Pode. On 16 Jan 2007, at 10:21 , Tim Panton wrote: On 15 Jan 2007, at 06:01, Tomer Horn wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) The nokia e60 is ok. (Much better than the original zyxel wifi phone) I found the configuration a struggle, but part of that was my fault, I couldn't see the difference between upper and lower case 'w's in the default nokia font! I was pleasantly surprised by it, but it is still a first generation solution, to be given to early adopters and technophiles. I lived with it for a week, and the only thing I can't cope with is that it isn't a clamshell. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Grandstream GXP-2000 and good support
Can you post the configuration template? --- Paulo. On May 18, 2005, at 20:22, Peter Svensson wrote: We recently purchased a Grandstream GXP-2000 phone and I would like to share our experiences with it, especially out very good support experience. The phone was easy enough to set up. The phone was configured using a configuration file served via tftp. Creating the configuration file was a bit tricky since no template was released for this particular phone. Most options could be set from the template for the Budgetone/Handytone products. Most features work as advertised. In speaker phone mode the microphone volume is too low and the phone needs an integrated acoustic echo canceler. The speaker itself is nice and clear and work very well when just listening in. During testing we noted a problem with one-way audio when calls were placed almost back-to-back to the phone. We notified Grandstream through an email and though no more of it. After 3 hours we received a request for a tcpdump log, after 6 hours we received a confirmation that the support personnel had replicated the problem and within 24 hours we received a new firmware correcting the fix! I cannot emphasize enough the impression such quick and professional support makes. Especially since the problem had a workaround (we found that pressing hold twice cleared up the problem). No nonsense questions about whether we tried rebooting the phone. The new firmware was a beta of their next firmware, I guess. Some new features were added like: * multiple accounts now with user selectable names * auto-answer selectable per account * better display texts * even more configurable options We also received a configuration template that allowed complete control over the phone from the server. With the next firmware the phone does feel ready for deployment in a corporation. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] zaphfc
Hello, Here it goes: zaptel.conf: --- span=1,1,3,ccs,ami bchan=1-2 dchan=3 --- zapata.conf --- switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local channel = 1 - Thanks, --- Paulo Loureiro. On Mon, 2004-04-19 at 21:27, Arnaud Pignard wrote: Hello, Can you post zapata.conf and zaptel.conf ? It's seems a config file problem. At 19:32 19/04/2004, you wrote: Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc
Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls
Well, after days chasing this ghost, the problem seems to be solved: - There was a problem with the NTP server (not responding to clients due to config restrictions) - upgrade gs firmware to latest available: 1.4.50. If the NTP server is down gs phones keep trying to reach it and drop some of the packets received from * during a call. * thinks the phone is dead and drops the call. Anyone can confirm this? --- Paulo Loureiro. On Fri, 2004-03-05 at 19:24, Bartosz Jozwiak wrote: I have couple of GS phone and CISCO 7960. The funny thing is that two of that GS phone keep disconnecting and also CISCO 7960 phone keeps disconnecting. But the problem appear month ago! This is really strange! Bart Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc: no channel
Hello, * installed with bri bri-stuff.0.0.2rc12 using a HFC based BRI board configured with zaptel and zaphfc (no errors on load). Whenever I try to make a call thru isnd (Zap/1/x), I get: app_dial.c:527 dial_exec: Unable to create channel of type 'Zap' show channels gives: Chan Extension Context Language MusicOnHold 1meridian-in 2meridian-in Any clues? --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dropped calls
Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dropped calls
Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
