Re: [asterisk-users] pulling my hair out over voicemail
On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL PROTECTED] wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: Maybe the SIP config is wrong? Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Can you places other calls from that new phone? Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. What version? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime device update weirdness
Hello,
We use Asterisk Realtime for our billing software. 200+ installations of
Asterisk with Realtime, but I see this for the first time.
Asterisk 1.4.17, Addons 1.4.5, No patches, no NAT - just plain simple
installation.
With debug I can see:
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:138 realtime_mysql: MySQL
RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = '109' AND host =
'dynamic'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 1 rows on table: devices
Notice update: UPDATE devices SET ipaddr = '0.0.0.0', port = '0', regseconds =
'0' WHERE name = '109'
Correct behaviour is: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
Why update to 0.0.0.0 is executed? It makes devices unreachable. When device
reregisters - it becomes available for short time - then again - update to
0.0.0.0. Why it is happening?
For temporaly solution i had to patch res_config_mysql.c at line 342, added
such lines:
if ((!strcmp(newparam, ipaddr)) (!strcmp(buf, 0.0.0.0))){
ast_log(LOG_DEBUG,MySQL RealTime: Avoided to update %s to %s
!!!\n, newparam, buf);
ast_mutex_unlock(mysql_lock);
return -1;
}
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
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[asterisk-users] Could not find ooh323.conf
Hi, I installed Asterisk, asterisk-addons, pwlib, h323plus,opal and gnugk. I am searching for /etc/asterisk/ooh323.conf. It is not there. Can anybody please tell me how to get ooh323.conf. Thanking you, Regards, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - SIP phones supporting LLDP-Med
Hi, Has anyone heard of SIP phones supporting LLDP-Med ? Mitel or Avaya phones are supposed to support it but I don't if it applies to SIP firmware enabled hardphones or not. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming call from SIP proxy to asterisk
Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as well as the media path. Client-1-- Asterisk- SIP Proxy Client-2-- So for call from client-1 to client-2, asterisk forwards the INVITE to the SIP proxy but when the SIP proxy gives the INVITE back to asterisk (as asterisk registered client-2 with the SIP proxy), asterisk is challenging the incoming INVITE. It seems asterisk is seeing the INVITE from SIP proxy as INVITE coming from client (this is I guess as per the lookup that asterisk performs). However is there a way to have it first match the INVITE host IP rather than FROM user first? Or rather is there a way to have this setup working? Regards, Mayur ___ -- 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] Could not find ooh323.conf
Hi, preeta. ppwc Hi, ppwc I installed Asterisk, asterisk-addons, pwlib, h323plus,opal and gnugk. ppwc I am searching for /etc/asterisk/ooh323.conf. It is not there. ppwc Can anybody please tell me how to get ooh323.conf. In source of asterisk-addons there is a file asterisk-ooh323c/h323.conf.sample Just copy to /etc/asterisk/ and rename to h323.conf -- Alexey Shimeshov mailto:[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
[asterisk-users] createlink with out agents in 1.4
Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [my-q] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm The calls are being recorded, but no entry appears in cdr (obviously). I can add the filename to userfield using Set(CDR(userfield)=filename), just before calling Queue. But file name will be present in all calls that entered the queue, the previous behavior was that only those calls which was actually connected to agents had this entry. That field was one easy way to find out which calls were connected to agents by looking at the cdr alone, and I am using this feature in a home brew call analysis software. I would be very happy if this feature can be emulated in 1.4 with out using agents channel. thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropped calls
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: zt_hangup(Zap/17-1) [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Hangup: channel: 17 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Updated conferencing on 17, with 0 conference users [Jan 31 07:10:43] VERBOSE[3131] logger.c: -- Hungup 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] app_dial.c: Exiting with DIALSTATUS=NOANSWER. [Jan 31 07:10:43] DEBUG[2695] devicestate.c: No provider found, checking channel drivers for Zap - 17 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] VERBOSE[3131] logger.c: == Auto fallthrough, channel 'SIP/dep2_1154-08202968' status is 'NOANSWER' [Jan 31 07:10:43] DEBUG[3131] channel.c: Soft-Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hangup call SIP/dep2_1154-08202968, SIP callid [EMAIL PROTECTED]) [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hanging up channel in state Ring (not UP) [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel SIP/dep2_1154-08202968 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag c136d668-768786 Our tag: as0bc591fc [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 2b4f6f33-768786 Our tag: as496fd97d [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 73176828-768785 Our tag: as1ab79f58 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag eae1f94d-768783 Our tag: as1b0024a8 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag f0629993-768783 Our tag: as3f520446 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 728b9929-768782 Our tag: as222bab2d Regards, McCoy ___ -- 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] Dropped calls
On Jan 31, 2008 6:45 AM, mccoy silva [EMAIL PROTECTED] wrote: I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: zt_hangup(Zap/17-1) [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Hangup: channel: 17 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Updated conferencing on 17, with 0 conference users [Jan 31 07:10:43] VERBOSE[3131] logger.c: -- Hungup 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] app_dial.c: Exiting with DIALSTATUS=NOANSWER. [Jan 31 07:10:43] DEBUG[2695] devicestate.c: No provider found, checking channel drivers for Zap - 17 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] VERBOSE[3131] logger.c: == Auto fallthrough, channel 'SIP/dep2_1154-08202968' status is 'NOANSWER' [Jan 31 07:10:43] DEBUG[3131] channel.c: Soft-Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hangup call SIP/dep2_1154-08202968, SIP callid [EMAIL PROTECTED]) [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hanging up channel in state Ring (not UP) [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel SIP/dep2_1154-08202968 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag c136d668-768786 Our tag: as0bc591fc [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 2b4f6f33-768786 Our tag: as496fd97d [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 73176828-768785 Our tag: as1ab79f58 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag eae1f94d-768783 Our tag: as1b0024a8 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag f0629993-768783 Our tag: as3f520446 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 728b9929-768782 Our tag: as222bab2d Regards, McCoy You need to Answer() the call in your dialplan, that is my guess without seeing your dialplan. Try adding EXTEN,1,Answer() before the rest of the stuff in your dialplan in the context that handles your inbound calls. Thanks, Steve Totaro ___ -- 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] Parking lot
pbx*CLI show application ParkAndAnnounce -= Info about application 'ParkAndAnnounce' =- [Synopsis] Park and Announce [Description] ParkAndAnnounce(announce:template|timeout|dial|[return_context]): Park a call into the parkinglot and announce the call over the console. announce template: colon separated list of files to announce, the word PARKED will be replaced by a say_digits of the ext the call is parked in timeout: time in seconds before the call returns into the return context. dial: The app_dial style resource to call to make the announcement. Console/dsp calls the console. return_context: the goto style label to jump the call back into after timeout. default=prio+1 On Jan 30, 2008 1:13 PM, Al lists [EMAIL PROTECTED] wrote: Is there any way to have Asterisk call an extension in dial plan instead of original extension after timeout? Like extension A puts the caller in parking lot, he leaves the phone and forgets about it, instead of having that phone rings after timeout, have a group of phones rings. ___ -- 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] Default delay time for Attended call transfer
First time or second time they hit transfer? Dial plan config? 2008/1/30 Don Smith [EMAIL PROTECTED]: Greetings, I have an issue with the length of time that passes from when someone hits the transfer soft key on a Cisco 7940, after doing an attended transfer, and when the recipient's connects with the transferred call. It appears to be around 6 seconds. Is there a .conf in Asterisk where this time can be reduced? Thank you for your help Don No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.16/1251 - Release Date: 1/30/2008 9:29 AM ___ -- 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] createlink with out agents in 1.4
On 1/31/08, Rajkumar S [EMAIL PROTECTED] wrote: Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [my-q] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm The calls are being recorded, but no entry appears in cdr (obviously). I can add the filename to userfield using Set(CDR(userfield)=filename), just before calling Queue. But file name will be present in all calls that entered the queue, the previous behavior was that only those calls which was actually connected to agents had this entry. That field was one easy way to find out which calls were connected to agents by looking at the cdr alone, and I am using this feature in a home brew call analysis software. I would be very happy if this feature can be emulated in 1.4 with out using agents channel. I think it's correct - if you set userfield - it will be written - no matter if call has been answered or not. If you would do ResetCDR before entering queue, you would have disposition=ANSWERED for queue calls that got answered by agents. Also you would have billsec= agent talk time and duration=total wait time in queue + talk time. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] pulling my hair out over voicemail
John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] pulling my hair out over voicemail
Try this: exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,VoiceMailMain() You do not specify the mailbox number in the call to the application. You only specify the number to VoiceMail() HTH, Shane On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote: John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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 -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ -- 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] Server Compatibility List for Asterisk
It is my understanding that the cast majority of the compatibility issues went away with the recent chipset change on the digium cards. Soa compatibility list really isn't needed. I've run the digium cards on all manner of Dell hardware (from old-school desktops all the way to the high end servers) and have never had issues. On 1/31/08, broadband Voice [EMAIL PROTECTED] wrote: Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Analog Adapters ?
I have a friend with a small business running a small SIP based phone system. He was looking into providing some SIP phones for a couple of remote teleworkers, but as he started to look around and ask me questions he ran across analog adapters which made him curious. He proceeded to ask me if there was an analog adapter that provided the following functionality in which my reply was simply, I don't know. I have NO experience with any analog adapters. I know that the basic function is simple, the adapter creates the SIP session if you will to the server. It then allows you to connect pretty much any analog device of your choosing. He however is wanting something that connects using both SIP to the server and PSTN. But his request does not stop there. He wants to be able to choose on the fly which SIP or PSTN connection he utilizes for any given outbound call the user makes. Basically, analog adapter connects to both voip pbx via sip, and PSTN. Analog phone connects to analog adapter. User picks up phone and could ideally press say 8 to make a call over the voip service or 9 to make a call over the attached PSTN. Sounds simple enough. And I know they do make adapters that connect to both a sip voip service and to the PSTN via a FXS port. Something like the Linksys SPA3102. However I am not certain that these devices allow for the individual to easily choose which service to use. I have to assume they do because well otherwise I have a hard time understanding how useful they would be otherwise. I notice a couple of the features listed stand out as possibly what they are looking for but any clarification from others with more experience and personal knowledge would be helpful. Features listed: Service Authentication via PIN, Digest, Caller ID (Bellcore Type 1) Per Call Authentication and Associated Routing Appreciate any responses. ___ -- 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] pulling my hair out over voicemail
Shane D wrote: Try this: exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,VoiceMailMain() You do not specify the mailbox number in the call to the application. You only specify the number to VoiceMail() HTH, Shane On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote: John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com What do you mean you do not use the mailbox in Voicemailmain see below: *CLI -= Info about application 'VoiceMailMain' =- [Synopsis] Check Voicemail messages [Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Options: p- Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller. g(#) - Use the specified amount of gain when recording a voicemail message. The units are whole-number decibels (dB). s- Skip checking the passcode for the mailbox. a(#) - Skip folder prompt and go directly to folder specified. Defaults to INBOX JohnM ___ -- 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] pulling my hair out over voicemail
Okay, What I ment was you don't have to. On 1/31/08, John Millican [EMAIL PROTECTED] wrote: Shane D wrote: Try this: exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,VoiceMailMain() You do not specify the mailbox number in the call to the application. You only specify the number to VoiceMail() HTH, Shane On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote: John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com What do you mean you do not use the mailbox in Voicemailmain see below: *CLI -= Info about application 'VoiceMailMain' =- [Synopsis] Check Voicemail messages [Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Options: p- Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller. g(#) - Use the specified amount of gain when recording a voicemail message. The units are whole-number decibels (dB). s- Skip checking the passcode for the mailbox. a(#) - Skip folder prompt and go directly to folder specified. Defaults to INBOX JohnM ___ -- 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 -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ -- 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] pulling my hair out over voicemail
Here are my configs: sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 context=default extensions.conf: [default] exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain Calling from phone to phone is fine, and inbound and outbound calling is fine. But when I call voicemail, I dont hear anything. When I view console in CLI I see this when attempting to dial the voicemail extension: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081d65c8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081d65c8, 2) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081d65c8, [EMAIL PROTECTED]) in new stack -- SIP/6001-081d65c8 Playing 'vm-login' (language 'en') [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: Couldn't read username Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE So it plays the greetings, and is working, I just cant hear it. -john On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote: On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL PROTECTED] wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: Maybe the SIP config is wrong? Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Can you places other calls from that new phone? Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. What version? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint is hanging when remote party ends call on hold
We are currently using Asterisk 1.4.9 with Unicall. We are experiencing an issue with hint hanging taking the extensions out of action until an asterisk restart. Details on this can be found at: http://bugs.digium.com/view.php?id=10474 We would like to upgrade Asterisk to 1.4.17 but are unsure how Unicall ties into Asterisk. Can we upgrade Asterisk and Unicall would be unaffected, or would we need to also upgrade and/or recompile Unicall and Zaptel? Thanks, Mark _ Mark S. Welch Network Administrator Sandler Travis Trade Advisory services, Inc. 248.474.7200 x1177 248.474.8500 (fax) [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.strtrade.com http://www.strtrade.com/ This is a transmission from Sandler Travis Trade Advisory Services, Inc. and is solely for the use of the intended addressee. It may contain information which is confidential and subject to attorney client privilege. If you are not the intended recipient please e-mail the sender and destroy all copies of this message and any attachment. Any unauthorized use of the contents of the message or attachments is strictly prohibited. ___ -- 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
On 1/31/08, 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?
You can't.
Retrieve the variable from inside AGI.
http://www.voip-info.org/wiki/view/get+variable
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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Re: [asterisk-users] Default delay time for Attended call
A call comes in from the PSTN, Asterisk answers it, it goes to the directory,
and then to the extension the caller designates and the user at that extension
answers. The user at the extension then wants to transfer the call to another
extension; on the Cisco 7940 they push the “more” soft key, then the “Transfer”
soft key, then enter the extension number they want to transfer to, and hit the
“dial” soft key. The user at the new extension answers and the talks to the
user doing the transfer. They agree to transfer the call to the new extension
and the person who got the original call then hits the “transfer” soft key and
hangs up. 6 seconds later the caller and the new extension can talk to each
other. The line at the new extension is silent for those 6 seconds.
Thank you for help.
Here is the extension.cong file:
[general]
static = yes
writeprotect = no
clearglobalvars = no
[globals]
CONSOLE = Console/dsp
[trunkint]
exten = _9011.,1,Dial(Zap/g1/${EXTEN:1},70,Tt)
[trunkld]
exten = _91NXXNXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1},70,Tt)
[trunktollfree]
exten = _91800NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91888NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91877NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91866NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[international]
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = trunklocal
include = trunktollfree
include = longdistance
include = parkedcalls
[macro-trunkdial]
exten = s,1,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[macro-stdexten]
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
w/ unavail announce
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy
announce
exten = _s-.,1,Goto(s-NOANSWER,1); Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
VoicemailMain
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call
screening ; option (or use
P for databased call screening)
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
w/ unavail announce
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy
announce
exten = s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a
polite Don't call again
script.
exten = s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a
telemarketer torture script.
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
VoicemailMain
[macro-page];
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for Jump and s is for ANY call
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA) ; This is for the PolyComs
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the
Grandstream, Snoms, and Others
exten = s,n,NoOp() ; Add others here and Post on the Wiki
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[default]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)
exten = _9911,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)
exten = 6500,1,VoiceMailMain
exten = 6500,1,NoOp(${Exten:6:11})
exten = 6500,2,VoiceMailMain(s${CALLERID(all):1})
exten = o,1,Goto(default,6000,1)
include = parkedcalls
;[asterisk_guitools]
;exten = executecommand,1,System(${command})
;exten = executecommand,n,Hangup()
;exten = record_vmenu,1,Answer
;exten = record_vmenu,n,Playback(vm-intro)
;exten = record_vmenu,n,Record(${var1})
;exten = record_vmenu,n,Playback(vm-saved)
;exten = record_vmenu,n,Playback(vm-goodbye)
;exten = record_vmenu,n,Hangup
;exten = play_file,1,Answer
;exten = play_file,n,Playback(${var1})
;exten = play_file,n,Hangup
;hasbeensetup = Y
[DID_trunk_1]
include = default
exten = s,1,Answer()
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Directory(default||f)
[support]
include = default
exten = _X.,1,Goto(default|6009|1)
exten = s,1,Goto(default|6009|1)
[numberplan-custom-1]
plancomment = DialPlan1
include =
Re: [asterisk-users] Meetme voice quality problems
On Jan 30, 2008 10:35 PM, Dan Austin [EMAIL PROTECTED] wrote: Franklin wrote: ztdummy can give you issues as a timing device. Yes and no. See below Any way you could try using a Digium card just as a timing device to see if this helps? Tomasz wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded? it is 1024 Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.4-r3748 Zaptel Echo Canceller: MG2 ztdummy: RTC rate is 1024 how can I increase it? I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Yes I have it set. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Users Conference Friday Feb 1st @ 12 Noon EST: Hosted IVR
Our guest is tomorrow Mobeen Khan is Chief Operating Officer of Metaphor Solutions who offer Plug Play IVR On-Demand http://www.metaphorivr.com Instructions to join the conference: http://VoipUsersConference.org IRC: freenode.net #voip-users-conference The weekly Friday Noon VoIP Users Conference call was originally the Asterisk Users Conference but for unknown reasons, changed its name and URL :) ___ -- 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] Server Compatibility List for Asterisk
Thanks. I am getting a dual 3.0Ghz 2950 III. On 1/31/08, Erik Anderson [EMAIL PROTECTED] wrote: It is my understanding that the cast majority of the compatibility issues went away with the recent chipset change on the digium cards. Soa compatibility list really isn't needed. I've run the digium cards on all manner of Dell hardware (from old-school desktops all the way to the high end servers) and have never had issues. On 1/31/08, broadband Voice [EMAIL PROTECTED] wrote: Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TDMXXB and Electronic Noises
I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling ___ -- 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] Meetme voice quality problems
On Jan 30, 2008 5:48 PM, Matthew J. Roth [EMAIL PROTECTED] wrote: Tomasz Zieleniewski wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? Tomasz, Have you run zttest on the system? It verifies the accuracy of your timing source. Digium recommends an accuracy of at least 99.98%. If your accuracy is less than that it's probably the source of your problem. ztttest results show value below 99,98: [EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5 Opened pseudo zap interface, measuring accuracy... 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4064.232 system clock sample intervals (49.612%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.240 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4064.232 system clock sample intervals (49.612%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.240 system clock sample intervals (50.003%) --- Results after 11 passes --- Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference: 49.931827 Luckily, it's a problem with multiple solutions. The following thread documents some kernel configuration changes that you can make to improve the quality of ztdummy as a timing source: Recommendations for kernel config http://lists.digium.com/pipermail/asterisk-users/2007-October/197778.html Do You know how can I check and set kernel timer frequency? My preferred solution is to use an empty TDM400P as a timing source. It will cost you a little bit of money, but it's an easy way to reliably solve your problem. You'll find a few posts about it if you search the list, but this one has most of the information you'll need: Empty Wildcard TDM400P as a MeetMe timer. http://lists.digium.com/pipermail/asterisk-users/2007-March/182005.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] Digium TDMXXB and Electronic Noises
On Jan 31, 2008 12:28 PM, Matthew Yingling [EMAIL PROTECTED] wrote: I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling IRQ issues i would suspect. Also,just because two machines are the same make and model absolutely does not mean that they have the same hardware. I have seen exact server models ordered from CDW at the same time have very different chipsets. Thanks, Steve Totaro ___ -- 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] pulling my hair out over voicemail
John Von Essen wrote: Here are my configs: [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 I believe you need to include a context on your mailbox line, such as [EMAIL PROTECTED] Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] pulling my hair out over voicemail
Very odd. Could you try taking the mailbox line out of sip.conf and see what happens? On 1/31/08, John Von Essen [EMAIL PROTECTED] wrote: Here are my configs: sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 context=default extensions.conf: [default] exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain Calling from phone to phone is fine, and inbound and outbound calling is fine. But when I call voicemail, I dont hear anything. When I view console in CLI I see this when attempting to dial the voicemail extension: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081d65c8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081d65c8, 2) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081d65c8, [EMAIL PROTECTED]) in new stack -- SIP/6001-081d65c8 Playing 'vm-login' (language 'en') [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: Couldn't read username Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE So it plays the greetings, and is working, I just cant hear it. -john On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote: On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL PROTECTED] wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: Maybe the SIP config is wrong? Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Can you places other calls from that new phone? Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. What version? ___ -- 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 -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
Howdy, Excuse the neophyte questions... I was wondering: (1) what's involved in setting up a call with encrypted media (I'm on a cable network and don't want my calls snooped); (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? (3) my PSTN service provider that I have SIP trunking to doesn't provide SMS service (yet or possibly ever)--is there a way to shop-out SMS for my associated numbers from someone else? I eventually hope to have a cell phone with dataplan only that I then do SIP-over-UMTS (like a Nokia E70 or E61i, for instance)... if I'm moving towards having a single number, then I figure text messaging should work regardless of the device I'm using (softphone on a laptop, VoWifi handset, or POE hardphone). Any service/any device, right? That's what it's all about. (4) I have an HP all-in-one officejet and a SPA-2000 (or is it an unlocked PAP2-NA?) that I can use for fax service, but was wondering if I could use my Asterisk box (it's a 400MHz Geode LX) as a fax server without too much impact. Thanks, -Philip ___ -- 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] Can't read environment variable
Uhhh... just
export HOSTNAME
should be enough once it's been set.
Joost Kuif | Mobillion wrote:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen
Verzonden: Wednesday, January 30, 2008 1:58 PM
Aan: asterisk-users@lists.digium.com
Onderwerp: Re: [asterisk-users] Can't read environment variable
On Wed, Jan 30, 2008 at 01:38:48PM +0100, Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives
the hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Is that variable set?
cat /proc/PID_OF_ASTERISK/environ | tr '\0' '\n' | grep ^HOSTNAME=
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[asterisk-users] Pros and cons of internal_timing
List users, A recent post on MeetMe timing mentioned the internal_timing option, which can be configured to have Asterisk asynchronously generate outgoing RTP when a timing device (ie. ztdummy) is available. This allows Asterisk to produce outgoing audio in situations where no incoming audio is arriving due to, for example, silence suppression. This seems like a major improvement to the core functionality of Asterisk, but a search of the lists, the wiki, and the book didn't produce much information about the option. We are a Business Edition shop running ABE-B, which is based on the 1.2 codebase. However, the 1.4-based ABE-C was recently released so I'm interested in the potential pros and cons of internal_timing. For instance, how much does turning on silence suppression on a phone typically lower the bandwidth requirements per call and what are the effects on features such as VAD and CNG? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer PS - I apologize for the threadjack, but it seems that my posts never make it to the list when I try to start a new thread. ___ -- 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] Meetme voice quality problems
Tomasz Zieleniewski wrote: ztttest results show value below 99,98: [EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5 snip --- Results after 11 passes --- Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference: 49.931827 This is the first thing I would address. Get that average to at least 99.98% and it's likely that your problem will go away. Do You know how can I check and set kernel timer frequency? You can check the timer frequency as follows: # grep -e ^CONFIG_HZ /boot/config-`uname -r` CONFIG_HZ_1000=y CONFIG_HZ=1000 Setting it requires configuring and rebuilding the kernel. Try setting CONFIG_HZ=1000 and checking the results of zttest. Depending on how new your kernel is there are more options, but this is a good place to start. I settled on using an empty TDM400P as a timing source, because it is a simple solution that just works. This may still be your best bet, but I'll defer judgment on that to the list because Asterisk has evolved quite a bit since I made that decision. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] Digium TDMXXB and Electronic Noises
I second that. IRQ issues are more than likely causing the problem. Check your interrupts and see if your TDM cards are sharing IRQs with any other devices. From past experience, I know we would get the same behavior when an analog card was sharing an IRQ with a storage controller. Any amount of disk activity would cause little blips and beeps in the audio stream. Make sure you have all extraneous unneeded devices turned off in the BIOS. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 31, 2008 11:34:18 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Digium TDMXXB and Electronic Noises On Jan 31, 2008 12:28 PM, Matthew Yingling [EMAIL PROTECTED] wrote: I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling IRQ issues i would suspect. Also,just because two machines are the same make and model absolutely does not mean that they have the same hardware. I have seen exact server models ordered from CDW at the same time have very different chipsets. Thanks, Steve Totaro ___ -- 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] pulling my hair out over voicemail
John Von Essen wrote: Here are my configs: sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 context=default extensions.conf: [default] exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain Calling from phone to phone is fine, and inbound and outbound calling is fine. But when I call voicemail, I dont hear anything. When I view console in CLI I see this when attempting to dial the voicemail extension: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081d65c8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081d65c8, 2) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081d65c8, [EMAIL PROTECTED]) in new stack -- SIP/6001-081d65c8 Playing 'vm-login' (language 'en') [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: Couldn't read username Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE So it plays the greetings, and is working, I just cant hear it. what's your voicemail.conf looks like? also check the file permission and make sure asterisk can read it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- 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] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
On Thu, 2008-01-31 at 10:45 -0800, Philip Prindeville wrote: (1) what's involved in setting up a call with encrypted media (I'm on a cable network and don't want my calls snooped); Using IAX, it's pretty simple. See http://www.voip-info.org/wiki/view/IAX+encryption (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf for the phone. As far as shared-line appearances go, the only way I know of is to use the SPA-932 sidecar with the SPA-962 phone, as documented here: http://www.voipinfo.org/wiki/view/SPA-962 -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] Digium TDMXXB and Electronic Noises
You are probably both correct. I noticed that both of our TDM cards, and the Ethernet card are all sharing the same IRQ. Since we do VOIP internally and analog externally, that IRQ is getting hit twice for any outbound or inbound calls. The system is new, and the OS supports ACPI, so I'm not yet sure what's going on. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Nelson Sent: Thursday, January 31, 2008 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium TDMXXB and Electronic Noises I second that. IRQ issues are more than likely causing the problem. Check your interrupts and see if your TDM cards are sharing IRQs with any other devices. From past experience, I know we would get the same behavior when an analog card was sharing an IRQ with a storage controller. Any amount of disk activity would cause little blips and beeps in the audio stream. Make sure you have all extraneous unneeded devices turned off in the BIOS. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 31, 2008 11:34:18 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Digium TDMXXB and Electronic Noises On Jan 31, 2008 12:28 PM, Matthew Yingling [EMAIL PROTECTED] wrote: I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling IRQ issues i would suspect. Also,just because two machines are the same make and model absolutely does not mean that they have the same hardware. I have seen exact server models ordered from CDW at the same time have very different chipsets. Thanks, Steve Totaro ___ -- 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
[asterisk-users] FS: A20101D Sangoma Board 2 Port FXO 2 Port FXS w/ Echo Can
Purchased a Sangoma board for a company that went defunct on 11/5/2007. Will accept $500 or best offer. Note this board does have echo cancelation in hardware. Will provide a copy of the receipt. Details A20101D Sangoma Board 2 Port FXO 2 Port FXS w/ Echo Can 20032D0-02679 FXS-03854 FXO-11991 Any questions, please let me know. Thanks! =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= Abel Computers ___ -- 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] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf for the phone Make sure also to add voice mail number under admin/advanced/Phone menu on the 942. It does not seem to pick it up automatically from asterisk like many phones do. If you’re configuring a lot of similar handsets, consider using an autoprovisioning script - it'll save you a hell of a lot of time in the long run. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
Using IAX, it's pretty simple. See http://www.voip-info.org/wiki/view/IAX+encryption Jared, perhaps you could clarify something on that voip-info.org article. It claims that encryption only works for auth=md5. Does that mean that using a public/private key for authentication (auth=rsa) will *not* result in encryption when encryption=yes is set? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem picking up a call with PickUpChan or PickUp
Hi, I have configured my SNOM 360 to monitor another extension by setting the following: [default] exten = user1,hint,SIP/user1 The next step was to define a function key on the phone as an extension with the value sip:[EMAIL PROTECTED] and later with sip:[EMAIL PROTECTED]|*8 When someone now calls extension 97 (which is the number of the corresponding phone), the LED on the SNOM 360 starts flashing. According to the documentation in the wiki (http://www.voip-info.org/wiki-Asterisk+phone+snom) I have tested a number of configurations on the phone, e.g. exten = _*8.,1,PickUpChan(SIP/user1) But all I get is a flashing LED, when I press the key, the display tells me connected, but the phone isn't connected and I don't get any message on the cli (debug level 5). Since I I' m using Asterisk 1.4.17-BRIstuffed-0.4.0-test6 Does anyone of you has a working configuration with SNOM phones that are able to pickup a call from a flasing LED? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alcatel omnipcx
Hi, can anyone tell me how i do a sip trunk between an asterisk and a alcatel omnipcx pbx with sip support tx, Pedro Santos ___ -- 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] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
Chris Bagnall wrote: (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf for the phone Make sure also to add voice mail number under admin/advanced/Phone menu on the 942. It does not seem to pick it up automatically from asterisk like many phones do. If you’re configuring a lot of similar handsets, consider using an autoprovisioning script - it'll save you a hell of a lot of time in the long run. Regards, Chris I've thought about doing that... My main two issues are: (1) it's a pain not being able to stuff an XML file via http into a Sipura (I think you either have to use TFTP or else HTTPS), and (2) not having a single source of configuration state to put into the phones. I could grab state out of multiple places, but besides being messy, that also leads to things getting into disagreement when they are edited in one place but not others, etc. I've also not found a good cheat sheet that says what fields need to be set to what. I can do the scripting/programming no problem. What I don't have time for is the learning curve of figuring out from scratch what settings work best. I figure someone else out there has already done that quite effectively. -Philip ___ -- 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] Incoming call from SIP proxy to asterisk
- Original Message From: Mayur [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, 31 January, 2008 9:59:42 AM Subject: [asterisk-users] Incoming call from SIP proxy to asterisk Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as well as the media path. Client-1à Asterisk---à SIP Proxy Client-2ß ß-- So for call from client-1 to client-2, asterisk forwards the INVITE to the SIP proxy but when the SIP proxy gives the INVITE back to asterisk (as asterisk registered client-2 with the SIP proxy), asterisk is challenging the incoming INVITE. It seems asterisk is seeing the INVITE from SIP proxy as INVITE coming from client (this is I guess as per the lookup that asterisk performs). However is there a way to have it first match the INVITE host IP rather than FROM user first? Or rather is there a way to have this setup working? Hi Mayur, Set the outboundproxy setting for the SIP accounts on Asterisk to the IP address of your SIP Proxy. Note there is also an outboundproxyport setting but it gets ignored by Asterisk. You won't need to worry about this if your SIP proxy is on port 5060. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Default delay time for Attended call
- Original Message From: Don Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, 31 January, 2008 4:46:27 PM Subject: Re: [asterisk-users] Default delay time for Attended call A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the “more” soft key, then the “Transfer” soft key, then enter the extension number they want to transfer to, and hit the “dial” soft key. The user at the new extension answers and the talks to the user doing the transfer. They agree to transfer the call to the new extension and the person who got the original call then hits the “transfer” soft key and hangs up. 6 seconds later the caller and the new extension can talk to each other. The line at the new extension is silent for those 6 seconds. Hi Don, There is no setting you can adjust on Asterisk. For an attended transfer there is no further interaction with the Asterisk dialplan once the transfer button on your Cisco is pressed it's all handled automagically in the SIP channel and channel.c. When I do an attended transfer through Asterisk the transfer takes about 1 second to complete, 6 seconds seems like a very large amount of time for the operation. With an attended transfer there are no new audio streams to set up for Asterisk and therefore no NAT that could be slowing things down as the audio trys to get through. The only thing that springs to mind would be if the REFER request from your Cisco was getting dropped along the way and taking a few re-transmits to get through. That wouldn't be likely to happen everytime though so a consistent 6 second delay is puzzling. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Analog Adapters ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 d4rk f1br wrote: He however is wanting something that connects using both SIP to the server and PSTN. But his request does not stop there. He wants to be able to choose on the fly which SIP or PSTN connection he utilizes for any given outbound call the user makes. Basically, analog adapter connects to both voip pbx via sip, and PSTN. Analog phone connects to analog adapter. User picks up phone and could ideally press say 8 to make a call over the voip service or 9 to make a call over the attached PSTN. Sounds simple enough. And I know they do make adapters that connect to both a sip voip service and to the PSTN via a FXS port. Something like the Linksys SPA3102. The SPA3102 will do that. It has dialplan functionality which will allow you to configure it the way you requested: 8 for VoIP, 9 for PSTN calls. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFHolktCFu3bIiwtTARAjr/AJ9zJmmhwORGVWc9f2eclT+RhI+H8ACgp97Z bFrYQBU/wZn5Yxy/rAoJNKk= =0BVS -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.18-rc4 Now Available
Asterisk 1.4.18-rc4 is now available. This release candidate includes an important fix for a regression related to the use of codec_g729 that caused decoders to not get properly released. Additional fixes added today that are included in this release candidate include: - fixes for some locking errors in chan_agent - a memory leak related to the use of AMI redirect - Solaris compatibility fixes - a fix related to call recordings from Monitor getting deleted before being mixed if a blind transfer is done from a Queue. Thanks to everyone that has jumped on to help out with testing of release candidates! It has already been extremely helpful. This release candidate is published for anyone that is interested in helping to test it for a couple of days before it is officially released. To download the release candidate, use the following svn command: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc4 If you would like it in tarball format, use the following commands: $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc4 $ tar -czvf asterisk-1.4.18-rc4.tar.gz asterisk-1.4.18-rc4/ Thanks! ___ -- 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] CallerID shows wrong values in manager interface
The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj Devraj Mukherjee wrote: CallerIDName: unknown State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Channel: SIP/103-098500d8 CallerID: 103 ___ -- 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] CallerID shows wrong values in manager interface
I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- 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] Analog Adapters ?
Barry L. Kline wrote: He however is wanting something that connects using both SIP to the server and PSTN. But his request does not stop there. He wants to be able to choose on the fly which SIP or PSTN connection he utilizes for any given outbound call the user makes. Basically, analog adapter connects to both voip pbx via sip, and PSTN. Analog phone connects to analog adapter. User picks up phone and could ideally press say 8 to make a call over the voip service or 9 to make a call over the attached PSTN. Sounds simple enough. And I know they do make adapters that connect to both a sip voip service and to the PSTN via a FXS port. Something like the Linksys SPA3102. The SPA3102 will do that. It has dialplan functionality which will allow you to configure it the way you requested: 8 for VoIP, 9 for PSTN calls. Barry I second that suggestion. I had a spare SPA3000 that I configured at the neighbor's house. Calls to my number and certain international destinations are routed through VOIP transparently. If you deploy this in a residential setting, I suggest picking out ONLY 9+ calls for VOIP and leaving 7 and 11-digit dialing alone. And BE VERY SURE that 911 calls will go through PSTN and promptly. And 9911 too, while you're at it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime warning
Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? ango ___ -- 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] Discover Asterisk 1.4 :: Jitterbug, no, Jitterbuffers
Johansson Olle E wrote: In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ I wrote a patch that lets you use the jitterbuffer in Asterisk 1.4 for more than just PSTN gateway functionality. Originally, there was no way to use it when connecting to Asterisk applications that did not create outbound channels and bridge calls (basically only Dial and Queue). This is actually still the case, but what I did was add support for using the jitterbuffer when you are bridged to a Local channel. That way, you can use it when connected to Voicemail, Meetme, or whatever else you want. See this post for more information: http://www.russellbryant.net/blog/index.php/2007/10/09/asterisk-jitterbuffer-support-for-applications/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] realtime warning
Rilawich Ango wrote: Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? Did you install res_config_mysql from asterisk-addons? -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h priority problem
I need to carry a variable over into the 'h' priority - so I can go back and clean up DB entries in a mysql database (time of call and so on) I tried using UNIQUEID but it seems that 'h' generates a new one. Anyone have any ideas? What can I use to carry a variable over into 'h'?? later, PaulH ___ -- 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] Problem with DTMF dialing
Sorry for taking so long to reply, This email got lost in translation, again. Ian Ian said the following on 30-Jan-08 03:57 PM Thaks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost routing. The problem I am having is dialing out using DTMF signalling. At the moment I am making do with Pulse dialing through the 3 analog lines. I can recieve calls on the Cellphone line without any problems, but cant dial out through it, as a cellphone cant do pulse dialing. I have run ztmonitor 1 -f gains, where 1 is the zap channel where the cellphone is located, while dialing the number 072 031 1294. I then went to audacity, on my own pc, and converted the raw file into mp3 format, mp3 is a compressed format, and hence may lose some quality. Generally you should stick with wav. ztmonitor should spit the appropriate sox command to do the conversion. Maybe it would look slightly different in the original format. Ok I tried this everywhich way I could but everytime I came up short of an answer. Meaning I am unable to find the right sox command to get this converted to wav on the same computer, so once again I got it to my pc, and then using my favourite friend, audacity I imported it as a raw format at 8000Hz, and exported it as a wav file this time, available for download from http://www.iancoetzee.za.net/gain.wav. it has the same effect, the numbers I dialed and the feedback I got is two different things. which is available for download at http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the playback I concluded that the DTMF signals being sent is totally wrong. Is that the whole tone? It is too short to be a valid DTMF. Yes that was the dial bit, this time I included the whole recording from beginning to end. if you count the tones you get to 10, which is the correct amount for South Africa. Another thing that got me worried is the fact that the last digit has a fair ammount of pause (about the same length of another tone) before it is sent. If you want I can upload the raw data to my server as well. Regards Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Technician Telephone : 012 664 2300 cellphone : 079 522 6519 Fax : 012 644 2902 E-mail : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Addons install success-Could not find ooh323.conf
Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing something wrong? What I should do to get ooh323.conf Thanking you, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] Analog Adapters ?
I have an SPA3000 and it works really great !!! It can do more than you say but "Per Call Authentication and Associated Routing", I dont understand what you mean. About your example with "press 8 ..." there are more eficient scenarios. You can can create a dialplan that automatically selects SIP or PSTN according to the destination number. At the same time you can eventually overlay that configuration with a prefix in order to select the wished route. Good Luck d4rk f1br escribi: I have a friend with a small business running a small SIP based phone system. He was looking into providing some SIP phones for a couple of remote teleworkers, but as he started to look around and ask me questions he ran across analog adapters which made him curious. He proceeded to ask me if there was an analog adapter that provided the following functionality in which my reply was simply, "I don't know". I have NO experience with any analog adapters. I know that the basic function is simple, the adapter creates the SIP session if you will to the server. It then allows you to connect pretty much any analog device of your choosing. He however is wanting something that connects using both SIP to the server and PSTN. But his request does not stop there. He wants to be able to choose on the fly which "SIP or PSTN" connection he utilizes for any given outbound call the user makes. Basically, analog adapter connects to both voip pbx via sip, and PSTN. Analog phone connects to analog adapter. User picks up phone and could ideally press say 8 to make a call over the voip service or 9 to make a call over the attached PSTN. Sounds simple enough. And I know they do make adapters that connect to both a sip voip service and to the PSTN via a FXS port. Something like the Linksys SPA3102. However I am not certain that these devices allow for the individual to easily choose which service to use. I have to assume they do because well otherwise I have a hard time understanding how useful they would be otherwise. I notice a couple of the features listed stand out as possibly what they are looking for but any clarification from others with more experience and personal knowledge would be helpful. Features listed: Service Authentication via PIN, Digest, Caller ID (Bellcore Type 1) Per Call Authentication and Associated Routing Appreciate any responses. ___ -- 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
