[asterisk-users] Fri March 12th @ 12 noon EST: SIP scanning, security and attacks + Hosted vs on-site voip
Hi all, Today's jam-packed sessions include the security theme for the first hour or so, then a debate about hosted vs local VoIP services. Hour one guests are Sjur Usken, telecom consultant who has been working with VoIP since 2002 and helping companies migrate to an all IP world and Sandro Gauci, a security researcher and consultant based in, author of VoIP security tools SIPVicious, VOIPPACK and VOIPSCANNER.com. They'll be talking about a number of realistic VoIP attacks and what's being exploited by fraudsters for profit. Hour two we expect Mike Oeth, Junction Networks CEO to join our regulars to talk about hosted vs local VoIP. There's also miscellaneous buzz about Android desk phones and such. Tune in, you won't regret it. General VUC info: http://vuc.me IRC anytime for info and during the calls: #vuc on Freenode.net or http://vuc.me/irc Start time for your time zone: http://vuc.me/next Access via SIP, Skype, PSTN, web widget and prayer pillows: http://vuc.me/how-to-participate/ To listen to a stream: http://vuc.me/talkshoe On March 26th, 2010 VUC will be be three years old! To celebrate, we're holding a 24-hour session to give everyone everywhere a chance to meet the VUC community. The discussion will be much wider than VoIP alone and there's no excuse for people of the Southern Hemisphere to miss this one! http://Voipathon.org for info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Time counting down and # detect
Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here
OMG I overlooked that portion Please honor my apology. On Thu, Mar 11, 2010 at 6:23 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 12 Mar 2010, Angelito Manansala wrote: If you are having trouble reading this email, read the online version http://now.eloqua.com/es.asp?s=491e=78675elq=55426a8b6c714f5bb6f2bf4b5d37bf55 . http://app.en25.com/e/er.aspx?s=491lid=215elq=55426a8b6c714f5bb6f2bf4b5d37bf55 Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release announcement. You are obliged, under the terms of your NDA with Digium, to keep this information confidential until the Switchvox SOHO 4.5 release is announced on March 30, 2010.* So am I missing something or did you just blatantly disregard the above warning to honor your NDA? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can not enable sip debug because CLI flooded
Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (31ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (39ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) I want to enable sip debugging but then my CLI is even more flooded with al the SIP OPTION packets... What can I do ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf payload 100
Probably has anyone idea how dtmf payload type could be changed in Asterisk say to 100? On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.comwrote: Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? Regards, Katerina -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf payload 100
12 mar 2010 kl. 10.45 skrev Katerina Borin: Probably has anyone idea how dtmf payload type could be changed in Asterisk say to 100? On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.com wrote: Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? If they have declared it correctly in the SDP, we will understand. Since Asterisk doesn't recognize the codec, I belive they have a bug in their system. In order for us to find out if Asterisk is doing wrong or if we can blame their system, we need to see the INVITE or 200 OK from their end. The information you have provided here is not enough. THanks, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: Can not enable sip debug because CLI flooded
Edit logger.conf and set the desired log level. To disable the messages below just remove the severity notice from console. console = notice,warning,error,debug Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di jonas kellens Inviato: venerdì 12 marzo 2010 10:20 A: Asterisk Mailing Oggetto: [asterisk-users] Can not enable sip debug because CLI flooded Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (31ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (39ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) I want to enable sip debugging but then my CLI is even more flooded with al the SIP OPTION packets... What can I do ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: Can not enable sip debug because CLI flooded
On 12 Mar 2010, at 10:05, Alexandru Oniciuc wrote: Edit logger.conf and set the desired log level. To disable the messages below just remove the severity notice from console. console = notice,warning,error,debug Because if you can't see it it's not broken? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: Can not enable sip debug because CLI flooded
That is indeed an option, thank you. It also went away by restarting Asterisk, but this is not desirable in production environment. On Fri, 2010-03-12 at 11:05 +0100, Alexandru Oniciuc wrote: Edit logger.conf and set the desired log level. To disable the messages below just remove the severity notice from console. console = notice,warning,error,debug Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec preference
If I have this is sip.conf : [general] disallow=all allow=g729 allow=alaw The prefered codecs set in my Grandstream phone is G729, alaw. In the sip peer definition I have commented out 'disallow=' and 'allow='. The prefered codecs set in the Zoiper softphone is alaw, gsm. In the sip peer definition I have commented out 'disallow=' and 'allow='. When making a call from the Grandstream to the Zoiper softphone, with Asterisk staying in the media path (canreinvite=no), you would expect all 3 of them to use alaw. But this is what happens : The Grandstream : v=0 o=test3 8000 8001 IN IP4 192.168.1.101 (-- Grandstream IP-address) s=SIP Call c=IN IP4 192.168.1.101 t=0 0 m=audio 10082 RTP/AVP 18 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 [Mar 12 11:44:14] Found RTP audio format 18 [Mar 12 11:44:14] Found RTP audio format 8 [Mar 12 11:44:14] Found RTP audio format 101 [Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082 [Mar 12 11:44:14] Found audio description format G729 for ID 18 [Mar 12 11:44:14] Found audio description format PCMA for ID 8 [Mar 12 11:44:14] Found audio description format telephone-event for ID 101 [Mar 12 11:44:14] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw| g729) [Mar 12 11:44:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082 [Mar 12 11:44:14] -- Called test1 (-- the Zoiper softphone) [Mar 12 11:44:14] -- Got SIP response 415 Unsupported Media Type back from 192.168.1.106 (IP-address of Zoiper softphone) So what happens here with the codec negotiation between Asterisk and the Zoiper softphone ?? Making the call the other way around (Zoiper calls Grandstream) the call succeeds and the codec is alaw... like it should be. I do get the warnings : [Mar 12 11:53:30] WARNING[23703]: channel.c:3340 ast_channel_make_compatible: No path to translate from SIP/test3-0a168b48(256) to SIP/test1-0a166d00(8) [Mar 12 11:53:31] -- SIP/test3-0a168b48 is ringing [Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Mar 12 11:53:34] -- SIP/test3-0a168b48 answered SIP/test1-0a166d00 [Mar 12 11:53:34] -- Packet2Packet bridging SIP/test1-0a166d00 and SIP/test3-0a168b48 (I know there are G729-licences to translate from G729 to alaw, but if both support alaw, then alaw should be the negotiated codec, no ?!) Can this be explained ? Thanks. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple RTP port ranges for SIP
Am 10.03.2010 17:33, schrieb Kevin P. Fleming: Klaus Darilion wrote: That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow configuration of different ranges for UDPTL and RTP (although it shouldn't be a problem to configure the same ports in rtp.conf and udptl.conf) It absolutely would be a problem to have identical, or even overlapping, port ranges specified in rtp.conf and udptl.conf. Those port numbers are UDP port numbers, and they must be unique across the system for things to work properly. Hi Kevin! Is Asterisk really that thumb and announces port befores testing if it actually can open the socket? Usually you have other services running on the same server to (e.g. DNS uses UDP ports), and just specifying port=1000-1999 in rtp.conf does not prevent that any other process on this server uses one of these ports. Thus, usually an application will try to open an UDP socket, and if it that fails, it just tries to open another one (with some logic behind). So, if port 1000 is already taken by another application, Asterisk should try to open another port. Thus I thought that if udptl opened a port within the portrange of rtp, res_rtp should be able to handle this. If this is not the case, IMO it is a serious bug in Asterisk. regards Klaus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses
Am 02.03.2010 13:29, schrieb Magnus Benngård: Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se Step 1 could be to send to the first ip/host and accept from both. Step 2 could be round-robin send if both are up and alive... IMO this would be a nice feature. Btw, did try trunk version, no support for multiple SRV records there. IIRC correctly there is a patch on the bugtracker for SRV handling, but I do not know if that patch would fix this too. regards klaus Am 02.03.2010 08:50, schrieb Magnus Benngård: Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Hi Magnus! Asterisk does not support multiple SRV records (expcet there were some recent changes which I missed) - it takes one of the most priors and use it all the time. Thus, in your scenario you have to specify the possible inbound sources manually as peers: [tele2-1] type=peer host=130.244.190.42 context=fromTele2 ... [tele2-2] type=peer host=130.244.190.46 context=fromTele2 ... regards klaus Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones won't stop ringing
Quoting Jason Aarons (US) jason.aar...@us.didata.com: I'm experiencing runaway ringing too, can we make this a class action against someone? Strangely enough, I have experienced this on what is a small domestic system - when a call is answered, sometimes other SIP phones (softphones only tried so far) keep ringing when the call is answered on a Zap or IAX phone - I think I have also had it happen when the answering phone was a SIP phone. It happens just occasionally, but can be a bit of a nuisance. Zap phones have never rung on and I can't remember it happening with an IAX phone. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses
12 mar 2010 kl. 12.01 skrev Klaus Darilion: Am 02.03.2010 13:29, schrieb Magnus Benngård: Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se Step 1 could be to send to the first ip/host and accept from both. Step 2 could be round-robin send if both are up and alive... IMO this would be a nice feature. Check my peerfailover branch. Btw, did try trunk version, no support for multiple SRV records there. IIRC correctly there is a patch on the bugtracker for SRV handling, but I do not know if that patch would fix this too. I haven't seen that. Interesting. /O regards klaus Am 02.03.2010 08:50, schrieb Magnus Benngård: Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Hi Magnus! Asterisk does not support multiple SRV records (expcet there were some recent changes which I missed) - it takes one of the most priors and use it all the time. Thus, in your scenario you have to specify the possible inbound sources manually as peers: [tele2-1] type=peer host=130.244.190.42 context=fromTele2 ... [tele2-2] type=peer host=130.244.190.46 context=fromTele2 ... regards klaus Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func odbc and mult iquery
2010/3/10 Tilghman Lesher tles...@digium.com
On Wednesday 10 March 2010 02:09:54 voipas wrote:
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both
tables. But asterisk only shows the first.
My database is MSSQL.
Yes, but only in 1.6.0 and above. You'll need to set mode=multirow in
func_odbc.conf, and the behavior of func_odbc changes dramatically. See
the sample func_odbc.conf for more information.
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Hi,
I'm using 1.6.0.20 asterisk and also I use multirow. I'm getting data like
this:
for (x=0; ${x} ${ODBCROWS}; x=${x} + 1) {
SET(ARRAY(variable,value)=${ODBC_FETCH(${RESULT})});
SET(${variable}=${value});
};
But using this, I can only retrieve first table. How to detect and retrieve
the second table?
Thanks
--
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Giedrius
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[asterisk-users] 1.2 to 1.6 and bristuff
Hi, I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap :) I was wondering if someone could point me at 3 things that I appear to have lost? 1) ZapEC(off) - Is there an equivalent dialplan command to request no EC on a channel before dialling in DAHDI? 2) rxfax(file.tiff) - I have found ReceiveFax(), but I am aware that much has happened in the faxing stakes recently, is there a good starting point to read about how this works in 1.6 with T.38 passthru/gateways etc? 3) Bristuff came with its own version of PickupChan() - Does anyone know if native Asterisk supports their Pickup mechanism? Or perhaps the Bristuff work has been ported to 1.6.2? I could not find it anywhere. Tzafrir used to run a GIT repo of that sort of work, but I have not dealt with it for so long I have lost the references to it :( If necessary, I can re-do the bristuff code that I need for 1.6.2, but thought I'd make sure I was not reinventing the wheel. Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 crash: gdb trace on core dump
Hi, I'm one of those people who still need to maintain * 1.2 systems and cannot easily upgrade. :-( My 1.2 systems were very stable until I upgraded from 1.2.37 to 1.2.40. I have made some changes within my dialplan but nothing unusual. Today I've had a crash: https://issues.asterisk.org/file_download.php?file_id=25571type=bug Yesterday I had another: https://issues.asterisk.org/file_download.php?file_id=25572type=bug Could anyone please have a look at these gdb traces? Other than that the traces seem to point to ast_expr2 and chan_iax2, I don't really have a clue as to why Asterisk crashes. Any ideas? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec preference
I would also add the following : sip.conf has : [general] disallow=all allow=g729 allow=alaw allow=gsm And again the same in the sip peer definition : disallow=all allow=g729 allow=alaw allow=gsm sip debug shows : [Mar 12 15:28:23] Found audio description format G729 for ID 18 [Mar 12 15:28:23] Found audio description format PCMA for ID 8 [Mar 12 15:28:23] Found audio description format GSM for ID 3 [Mar 12 15:28:23] Found audio description format telephone-event for ID 101 [Mar 12 15:28:23] Capabilities: us - 0xa (gsm|alaw), peer - audio=0x10e (gsm|alaw|g729)/video=0x0 (nothing), combined - 0xa (gsm|alaw) On an outgoing call from my Grandstream (with codecs G729, PCMA, GSM) to Asterisk. What happened here with the G729 codec ?? Using Asterisk 1.4.25.1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Regarding - P-Asserted identity and Privacy - SOLVED
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten = _1NX,n,Set(PRIVACY=${SIP_HEADER(Privacy)})
exten = _1NX,n,ExecIf($[${PRIVACY} = id]|SetCallerPres|prohib)
This makes the calls with privay ON sent as anonymous at the other end. One
more thing is to make sure you enable usecallingpres=yes in chan_dahdi.conf.
Thank you
Sandesh
On Fri, Mar 5, 2010 at 11:18 AM, das sandesh sandesh...@gmail.com wrote:
Hi All,
We have two servers, one server (SIP asterisk server) sending calls to the
second server(has PRI) which goes our through the PRI's (using TE 412p).
When the pprivacy is enabled: P-Asserted-Identity Header, privacy id are
sent in the header of SIP invite packet to the second server, how can we
identify this privacy and block the callerid as the call goes to the second
server which has the PRI cards (TDM circuit)? I tried setCallerPres(prob)
but it prohibits all calls, is there any way of identifying the calls with
the privacy ON coming from the first server and then block only those calls?
Server details:asterisk: 1.4.26.2
dahdi: 2.2.0.2
libpri: 1.4.10.1
Thanks for your help.
Regards
Sandesh
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[asterisk-users] ExtenSpy Problem
Hi I'm trying to get ExtenSpy to work but it wont, I'm dialling a number from my mobile which comes into our server and answering the number on a particular SIP extension which all works fine. I'm then dialling an exten from my own SIP extension which executes the ExtenSpy for the correct extension but I hear nothing. Here is the output in the CLI -- Executing Goto(SIP/hidden-081aba30, pack-hidden|s|1) -- Goto (pack-hidden,s,1) -- Executing NoOp(SIP/hidden-081aba30, ) -- Executing Wait(SIP/hidden-081aba30, 2) -- Executing Set(SIP/hidden-081aba30, CALLERID(num)=hidden) -- Executing Set(SIP/hidden-081aba30, CALLERID(name)=Ish Test) -- Executing Dial(SIP/hidden-081aba30, SIP/PACK504|30) -- Called PACK504 -- SIP/PACK504-081a6b18 is ringing -- SIP/PACK504-081a6b18 is ringing -- SIP/PACK504-081a6b18 is ringing -- SIP/PACK504-081a6b18 answered SIP/213.166.5.133-081aba30 -- Packet2Packet bridging SIP/213.166.5.133-081aba30 and SIP/PACK504-081a6b18 -- Executing ExtenSpy(SIP/PACK501-081a80a8, pack...@pack-local|bq) == Spawn extension (pack-local, 5504, 1) exited non-zero on 'SIP/PACK501-081a80a8' PACK504 does exist under the pack-local context and I get the same thing if I leave out the context part. I get the same thing whether I put in the b option or not and if I don't put in the q option I get the following. Also, you can see ExtenSpy being executed for the same extension that has answered the call. -- Executing ExtenSpy(SIP/PACK501-081acfe0, PACK503) -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') Does anyone have any thought/experience of this? Also, if a call is already being recorded by MixMonitor, can it also be spied on? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
Hi Zoa, Nice work. 1. It would be nice to have the T30 libraries and include files being distributed to the root filesystem or externally defined DESTDIR. 2. Do you have any plans to put the libt30 integration automated via the configure script in Asterisk ? In addition, just playing around with the T.38 Gateway functionality shows that it works fine with ecm disabled and speed 9600. However, I noticed few things: 1. If it is the ingress side (sending gateway) then it doesn't negotiate the 14400 even though originating fax device supports it, it always starts with 9600 speed. 2. ECM on capability seems to have an issue. If ecm is enabled on both the fax machines, then it gets disabled internally by the gateway. Changing the code to enable it all the time would fail the fax as well. Kind Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t38 ATA
Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple RTP port ranges for SIP
Klaus Darilion wrote: Is Asterisk really that thumb and announces port befores testing if it actually can open the socket? No. Usually you have other services running on the same server to (e.g. DNS uses UDP ports), and just specifying port=1000-1999 in rtp.conf does not prevent that any other process on this server uses one of these ports. Thus, usually an application will try to open an UDP socket, and if it that fails, it just tries to open another one (with some logic behind). So, if port 1000 is already taken by another application, Asterisk should try to open another port. Thus I thought that if udptl opened a port within the portrange of rtp, res_rtp should be able to handle this. If this is not the case, IMO it is a serious bug in Asterisk. It will. However, if you are using both RTP and UDPTL and have configured them for minimally-sized port ranges based on your expected traffic, and also used overlapping ranges, it would be easy for calls to fail because there are no port numbers available. Using non-overlapping ranges will make this much less likely. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func odbc and mult iquery
On Friday 12 March 2010 05:55:33 voipas wrote:
2010/3/10 Tilghman Lesher tles...@digium.com
On Wednesday 10 March 2010 02:09:54 voipas wrote:
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both
tables. But asterisk only shows the first.
My database is MSSQL.
Yes, but only in 1.6.0 and above. You'll need to set mode=multirow in
func_odbc.conf, and the behavior of func_odbc changes dramatically. See
the sample func_odbc.conf for more information.
I'm using 1.6.0.20 asterisk and also I use multirow. I'm getting data
like this:
for (x=0; ${x} ${ODBCROWS}; x=${x} + 1) {
SET(ARRAY(variable,value)=${ODBC_FETCH(${RESULT})});
SET(${variable}=${value});
};
But using this, I can only retrieve first table. How to detect and retrieve
the second table?
I haven't the faintest idea how returning two resultsets from a stored
procedure could possibly work, let alone how to retrieve it properly.
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Re: [asterisk-users] R: Can not enable sip debug because CLI flooded
On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens jonas.kell...@telenet.bewrote: Are you using SIP realtime? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t38 ATA
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex The Grandstreams do T.38 quite well, and they do support ECM. It is probably your service provider which is blocking ECM. Many of them do. People complain a lot about Grandstream, but its mostly their phones. Their ATA are amongst the better ones. Sadly, that doesn't mean an awful lot, as most ATAs are quite nasty. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add custom CDR fields to MySQL
Hi Alex,
I'm having the same problem and there is an open problem report about
it. However, if you modify your /etc/asterisk/cdr_custom.conf file to
add the field it will show up in your Master.csv log file but still not
in the DB record. You could in essence use the log file to rebuilt your
cdr DB through an external script which you could call through cron or
possibly at the end of you context inside of asterisk. I haven't written
the script yet but I bet someone out there has.
Regards,
Mike
Emanuele Carbone wrote:
Hi,
i think that you should modify the cdr_addon_mysql module, otherwise
you can add it in the userfield.
2010/3/11 Alejandro Recarey alexreca...@gmail.com
mailto:alexreca...@gmail.com
Hi all,
I've been trying to add a custom mysql field to my CDR's, but I must
be doing something wrong.
I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add:
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
This extension is executed, I can see it in the asterisk console.
I have added a new column in my MySQL database called q931. However,
the new field does not show up in my database or in the Master.csv
file.
Any help would be greatly appreciated.
Regards,
Alex
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Re: [asterisk-users] MWI and 1.6.1
Hi Dave, Thought I'd give you an update - I completely rebuilt my astdb the other night by renaming it, having * recreate it and then re-creating all my custom entries in it. Didn't have any effect, I had somebody report false MWI notifications again earlier this morning. -- Matt On Tue, Mar 9, 2010 at 11:34 PM, Matt Watson m...@mattgwatson.ca wrote: Hi Dave, Sure enough my astdb does contain references to VM files as shown with strings - doing the database dump however does not show the references. I'm not sure about the internals of how Berk DB works, however I;m also seeing references to lots of other data that really shouldn't be part of my config anymore either - like I can see some employee names that are no longer a part of our company and thus have been deleted from our * config, some several years ago. I suspect that berkdb is just not overwriting some of the data for whatever reason and has some internal mechanism for knowing what to ignore. I believe I can probably test your theory tomorrow evening though, I don't think I have too much in my astdb that can't be easily re-created, I think I can probably delete my astdb entirely and regenerate it. I'll just need to take a closer look at it first though. I would however like to believe that if * is no longer supposed to be using berkdb for any VM reference data, that any calls to read the voicemail counts from the DB should have been removed. -- Matt On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.comwrote: So a couple of questions I have for you Matt... If you run strings on your astdb file are you seeing references to messages files in it? #strings /var/lib/asterisk/astdb | grep -i msg and if so... If you run a db_dump185 on your astdb file do the references go away? #db_dump185 -p -f /tmp/astdb.dump astdb -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t38 ATA
Steve- On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. Ive tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex The Grandstreams do T.38 quite well, and they do support ECM. It is probably your service provider which is blocking ECM. Many of them do. I've heard that Grandstream uses TI Telogy DSP devices, which might help explain their reliable operation at T.38 data level. Telogy got started in early 1990s and was one of the first outfits to make it big in VoIP. They were acquired by TI in 1998 for about 850m. During the late 1990s period, Telogy did a lot of work on FoIP, including helping to write RFCs, contributions to ITU study groups, etc. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: t38 ATA
Hi Steve, the remote device is an Hylafax Server that does ECM. The sending fax device, that's attached to the ATA, is a Philips fax machine with ECM enabled. If I send with the same machine but attached to a Patton 4114 with T38 enabled my faxes go to the other end with ECM enabled and with no errors. I've also tested the Grandstream ATA with analog fax machines attached to the PSTN and the results are the same: bad quality faxes or no faxes at all. Again, tested with the Patton device I get no errors. In this case I think it's the Grandstream. Best Regards, Alex Da: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] per conto di Steve Underwood [ste...@coppice.org] Inviato: venerdì 12 marzo 2010 18.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] t38 ATA On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex The Grandstreams do T.38 quite well, and they do support ECM. It is probably your service provider which is blocking ECM. Many of them do. People complain a lot about Grandstream, but its mostly their phones. Their ATA are amongst the better ones. Sadly, that doesn't mean an awful lot, as most ATAs are quite nasty. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom not updating the directory list
Hi, I have a strange problem with all of our Polycom 550 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The part that doesn't work is the MAC-directory.cfg. If I make an update to this file and reboot the phones, they do not reflect the new directory list. The only way I was able to get the phone to see the new directory list was to Format the phone. Of course this is not the ideal way. Also to add, the MAC-directory.cfg files point to 0-directory.xml. This way I only have one file to maintain. Anyone knows why it's not pull the new MAC-directory.cfg file. Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem config pots documentation
hi, i'm looking for documentation on configuring asterisk to work with a modem that should work with an analog line. i don't see the info in the handbook or reference manual or o'reilly's. any references and/or links, much appreciated. thanks. g. _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing chan_H323 by yum?
We have a client with Asterisk 1.6 installed via yum (onto Centos). It did not included the chan_h323 driver apparently, so we installed add-ons by yum. We then got ooh323. Is it possible to install the H.323 drivers without compiling from source? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.
I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and rfc2833). Could someone with a similar configuration as mine verify if i have found a bug or not? Some system info: Asterisk 1.6.2.5 built by root @ XX on a x86_64 running Linux on 2010-03-02 20:15:09 UTC Skype For Asterisk Components: Channel Driver: 1.6.2.0_1.0.9.2 Library: 1.6.2.0_1.0.9.2 //Joakim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.
Joakim Eriksson wrote: I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and rfc2833). This is a known issue with SkypeIn and SkypeOut and is being addressed. There should be a Skype For Asterisk release soon that contains the changes required on its send; there are also changes being made in the SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.
Thank for the help :) Then i can just hope it gets fixed soon. (But now that i know about it, its not as critical anymore). //Joakim On Mar 12, 2010, at 8:24 PM, Kevin P. Fleming wrote: Joakim Eriksson wrote: I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and rfc2833). This is a known issue with SkypeIn and SkypeOut and is being addressed. There should be a Skype For Asterisk release soon that contains the changes required on its send; there are also changes being made in the SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_confbridge production ready?
I do not use ConfBridge() in a large installation. I use MeetMe on 1.6.0.* The timing is different for ConfBridge, as it does not require DAHDI. If you have that good of an experience with 1.4, why change anything? I like new things. ConfBridge eliminates the need for an external timing source (like the Sangoma card) which allows me to run Asterisk on our preferred OS, Solaris. It also supports 16kHz audio which fits in nicely with all my Polycom wideband phones. Unfortunately I answered my own question by installing Asterisk 1.6.2.x on solaris 10 and giving it a shot. Launching ConfBridge segfaults asterisk everytime :( Thanks for the feedback. Bob -- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.18 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Make sure to clear red alarm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * Fix problem with duplicate TXREQ packets in chan_iax2. (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) * Update documentation to not imply we support overriding options. (Closes issue #16855. Reported by davidw) * Modify queued frames from Local channels to not set the other side to up. (Closes issue #16816. Reported, tested by jamhed) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.18 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.6 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.6 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Make sure to clear red alarm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * Fix problem with duplicate TXREQ packets in chan_iax2 (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) * Fix crash in app_voicemail related to message counting. (Closes issue #16921. Reported, tested by whardier. Patched by seanbright) * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts (Reported, Patched, and Tested by alecdavis) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) * Fix ConfBridge crash when no timing module is loaded. (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.30 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.30. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.30 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow parallel make (-j) to work properly. 1.4 changes are quite different from the others. (Issue #16489. Reported by Chainsaw. Tested by Qwell) * Fix a memory leak in pbx_spool when using SetVar in a call file. (Closes issue #16554. Reported, tested by mav3rick. Patched by seanbright) * Fix bug with channel receiving wrong privileges after call parking. (Closes issue #16429. Reported, patched by Yasuhiro Konishi. Tested by dvossel) * Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. (Closes issue #16834. Reported, patched by kebl0155) * Remove color code sequences from verbose messages that go to logfiles. (Closes issue #16786 For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.30 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.26 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.26. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.26 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Make sure to clear red alarm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. (Closes issue #16683. Reported by wdoekes) * Resolve a crash caused by a race condition in app_chanspy.c (Closes issue #16678. Reported, patched by tim_ringenbach. Tested by dvossel) * Fix deadlock in app_queue with use_weight during reload. (Closes issue #16677. Reported, patched by tim_ringenbach) * Stop playing the message number multiple times in app_voicemail. Also remove some accidentally duplicated code, which may have been causing a memleak. (Closes issue #16579. Reported by kue. Patched by hokie21) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.26 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
i gave up on ReceiveFAX and uses iaxmodem/hylafax instead. Tommy Botten Jensen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Edwin Lam skrev: Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] I am running Asterisk 1.6.2.1 and 1.6.2.2 with Spandsp 0.0.6-pre17 with the exact same error messages. But my asterisk does not have any trouble apart from the messages themselves. I did however run into this earlier, and I believe it was fixed for the 1.6.2.x series. At least it worked for me. Best regards, Tommy Botten Jensen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuXYlwACgkQ573V05EH/pZpZQCeO8FPGqAJ4cRDlnyZOERbgNoj 0TEAmgOiY0byfIy3SIM5GR9gDrG+LZEY =oN/L -END PGP SIGNATURE- -- Edwin Lam edwin@officegeneral.com 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: Can not enable sip debug because CLI flooded
Yes I am for most of my SIP peers. On Fri, 2010-03-12 at 10:51 -0600, Warren Selby wrote: On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens jonas.kell...@telenet.be wrote: Are you using SIP realtime? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up RTP to flow between endpoints directly bypassing Asterisk
Hello, http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing through a SIP+rtpproxy server that can do transcoding ? Thanks and Regards, Vikram. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
Hi Gordon, What i'm doing now is that something like karaoke. While music is playing back, caller voice is being record by the way i mentioned earlier. I should give you the whole picture of what i'm doing. I did google for it, and Monitor() function seem to be the best choice to do that. I would prefer using Record() if somehow i could play back music while recording. Thanks, Quyps On Fri, 2010-03-12 at 10:43 +, Gordon Henderson wrote: On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
Here again, the script should be described as - Caller call to a number - Asterisk answer, play back music and start MONITORING as following + User press * to start MONITORING + Record is finished if: + User press # + OR message duration reach 60 second + Hangup Quyps On Sat, 2010-03-13 at 08:36 +0700, Pham Quy wrote: Hi Gordon, What i'm doing now is that something like karaoke. While music is playing back, caller voice is being record by the way i mentioned earlier. I should give you the whole picture of what i'm doing. I did google for it, and Monitor() function seem to be the best choice to do that. I would prefer using Record() if somehow i could play back music while recording. Thanks, Quyps On Fri, 2010-03-12 at 10:43 +, Gordon Henderson wrote: On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t38 ATA
On 03/13/2010 02:03 AM, Jeff Brower wrote: Steve- On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex The Grandstreams do T.38 quite well, and they do support ECM. It is probably your service provider which is blocking ECM. Many of them do. I've heard that Grandstream uses TI Telogy DSP devices, which might help explain their reliable operation at T.38 data level. Telogy got started in early 1990s and was one of the first outfits to make it big in VoIP. They were acquired by TI in 1998 for about 850m. During the late 1990s period, Telogy did a lot of work on FoIP, including helping to write RFCs, contributions to ITU study groups, etc. I think you will find Grandstream uses TI silicon, but all their own software. T.38 is an awful loose spec, so different implementations tend to behave quite differently. The flavour of a Grandstream packet exchange doesn't appear to be the same as a Telogy exchange. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype for Asterisk and regular expressions
Is there something strange about using regular expressions in the context to which incoming Skype calls go? If I set up accounts, foobar1, foobar2, etc, it doesn't seem to work to have: exten = _foobarX,1,... should it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
On 03/09/2010 07:31 AM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 Crashes of this kind are not uncommon, but the causes are: - Multiple versions of libtiff installed in different directories - Multiple versions of spandsp installed in different directories - Asterisk was built against a spandsp installed in a directory that is not in the library search path, while another version of spandsp is in a directory that is in the library search path. so, at run time the wrong library is picked up. Many machines will happily build and install a library to /usr/local, and then successfully like applications against it, even though /usr/local is not in the library search path. Dumb, but true. The installation information page for spandsp, at http://www.soft-switch.org/installing-spandsp.html , warns about these issues. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDILOOKUP doesn't return record
Hi All,
Found an issue with DUNDILOOKUP function in Asterisk 1.6.0.5.
I was using DUNDIQUERY (Set(ID=${DUNDIQUERY(${MNUM},priv,b)})) for
dundilookup and it was working fine.
But when I tried to use DUNDILOOKUP function
(Set(DL=${DUNDILOOKUP(${MNUM},priv,b)})), it didn't retuen me a
result. Moreover, the cli command 'dundi lookup 12...@priv' returned
me the result at the same time!
I also checked that ${MNUM} is set properly.
What can be a problem?
Please guide me where I do a mistake.
--SM
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