Re: [asterisk-users] agi playback to execute say.conf settings
Hi all, I am able to understand your solutions. Depending upon the india number reading method, I changed number reading setting in say.conf language. For more details visit my blog http://asterisknumbertovoice.blogspot.com/. It is working well with playback(num:123456,say) when I specified it in dialplan. Thanks, Ashik On Thu, Sep 2, 2010 at 7:08 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asteriskguru asteriskguru *Subject:* [asterisk-users] agi playback to execute say.conf settings Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. snip but when I write it in agi does not working. Here is agi debug output from asterisk. SIP/6000-000aAGI Rx EXEC playback num:333456,say -- AGI Script Executing Application: (playback) Options: (num:333456,say) SIP/6000-000aAGI Tx 200 result=0 Anybody have any ideas to work it out in agi playback ? Replace playback “num:334456,say” with “say number 334456” Refer to http://www.voip-info.org/wiki/view/say+number -- _ -- 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] Asterisk 1.8.0-beta4 Now Available
At 08:35 AM 8/24/2010, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4. I've now tried all the V1.8 betas including this and I always get a message telling me to read sip-retransmit.txt when I make a call from a SIP phone, Aastra480i out a DAHDI line on a Digium TDM-400 with 4 red cards back to one of the other lines. It rings once and then I get 4 or so of that message and it goes to voicemail. Soon as I go back to the latest 1.6 it works perfectly again. I've read the document many time and I have no clue what to do with the information. I only have one box to test on so I just test it the occasional quiet evening by making that one call and it always fails with these message. I have no idea what to do to try and make it work or if it's likely a bug or an error in my configuration. I got no errors in loading that should have any effect on this. So this is a show stopper for me and while I'd love to help test 1.8, I can't successfully make one call with it up. Any suggestions on what I might try to improve things? [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 3038c0be7937f81a5a5187441e4b3...@192.168.2.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Hanging up call 3038c0be7937f81a5a5187441e4b3...@192.168.3.235:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 4427982a1e30f2e06aded749152d4...@192.168.3.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Hanging up call 4427982a1e30f2e06aded749152d4...@192.168.3.235:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 0226b2630404b2eb3aa8a5eb789e9...@192.168.3.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. -- _ -- 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] 3Com 3102 Phones
On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they are useless without a NBX or NCP ? Any specs on how to configure linux to act like one? Thanks in advance Does anyone have a packet capture they can share of a 3Com phone registering and connecting the an NBX or NCP So I could at least see a full traffic connection not just a 0x8838 outbound packet Thanks -- _ -- 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 Fax
Hi I know that this topic was on the list maybe dozen of times. But I have a question regarding the fax support in asterisk, because all the information I could get does not give me the clear view of if. I read that Asterisk 1.8 will have strong fax (t.38) support, but I want to know if these four scenarios will be possible to achieve: fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- SPA2102 ATA --- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- PSTN --- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- IAX --- another Asterisk 1.8 --- SPA2102 ATA--- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- SIP --- another Asterisk 1.8 --- SPA2102 ATA --- fax machine For last three scenarios Asterisk should work as fax T.38 gateway. Is it possible? Cheers Andrew -- _ -- 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] SMS and fixed land lines
Hi, 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? 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] Is it possible to keep both call legs live after Dial()
Hi folks, After a fairly extensive Google trawl, I don't think the following is possible but would appreciate confirmation from anyone else who has tried something similar. I have an AGI (not particularly relevant) which is executed when someone calls into a specific extension. This AGI finds a suitable 'agent' (not actually a queuing system in the Asterisk Queue sense) and Dial()s this agent bridging the call. Now, ideally, I would be able to act on a 'decision' from a DTMF sequence from the agent's handset. I don't think this is possible unfortunately. Please correct me if I'm wrong. I can get a 'decision' from the agent by using the 't' Dial() option and have the agent key an extension corresponding to a 'decision'. This will suffice. From this I can call another AGI for the caller and continue processing them. I'd like to be able to play some audio to the agent and even let the agent call continue with another AGI. This bit I don't think is possible either? Thanks and kind regards, Barry -- _ -- 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] Macro when calling cellphone (GSM) + silence when connecting
Hello list,
I'm using the following macro when calling an external callphone/GSM
number :
[macro-press1]
exten = s,1,NoOp()
exten = s,n,Playback(/var/lib/asterisk/sounds/prompts/press1)
exten = s,n,Read(INPUT,,1,1,1)
exten = s,n,NoOp(input : ${INPUT})
exten = s,n,GoToIf($[${INPUT}==1]?exit:hangup)
exten = s,n(exit),NoOp(call accepted)
exten = s,n,MacroExit()
exten = s,n(hangup),Set(MACRO_RESULT=CONTINUE)
exten = s,n,NoOp(macro_result in macro : ${MACRO_RESULT})
exten = s,n,MacroExit()
The dialplan :
exten = s,n,Dial(${TRUNKOUT}/${TEL},,M(press1))
So the calling party and the called party are only connected together
when the called party presses 1 to accept the call.
When playing the prompt Press 1 to accept the call, the calling party
here's a silence (ringtone stops).
How can I have the ringtone be played untill the calling party and the
called party are effectively connected together ?!
I guess by calling the Playback-command, the call is answered. But that
means that the ringtone stops. While the called party still needs to
acknowledge the call.
Anyone has a solution ?!
Kind regards,
Jonas.
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Re: [asterisk-users] SMS and fixed land lines
Hi! 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? Yes, typically there is only one SMSC that can send you SMS on a fixed line; look at its Caller ID to identify a SMS call. Philipp -- _ -- 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] SMS and fixed land lines
2010/9/6 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? Yes, typically there is only one SMSC that can send you SMS on a fixed line; look at its Caller ID to identify a SMS call. Even when the call is coming from a cellphone ? Philipp -- _ -- 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] Possible malformed G729B - SID (VAD/DTX)frames from carrier endpoint ?
Steve- On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. We have Kamailio + rtpproxy running between the endpoints. Do you think it's reasonable to convert the first malformed SID frame (10 bytes) to 2 bytes, and then strip the following malformed SID frames until we see the talkspurt marker bit is set? We could do that... I'm wondering if anyone has seen such malformed SID frames before. As a couple of additional notes, between us and the remote endpoint there appears to be using an ALOE Systems (formerly MERA systems) MSiP system. So far the SDP negotiations we've tried (e.g. a=fmtp:18 annexb=no) have not convinced the remote endpoint to disable VAD. -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
Re: [asterisk-users] Possible malformed G729B - SID (VAD/DTX)frames from carrier endpoint ?
On 09/06/2010 11:18 PM, Jeff Brower wrote: Steve- On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. We have Kamailio + rtpproxy running between the endpoints. Do you think it's reasonable to convert the first malformed SID frame (10 bytes) to 2 bytes, and then strip the following malformed SID frames until we see the talkspurt marker bit is set? We could do that... I'm wondering if anyone has seen such malformed SID frames before. As a couple of additional notes, between us and the remote endpoint there appears to be using an ALOE Systems (formerly MERA systems) MSiP system. So far the SDP negotiations we've tried (e.g. a=fmtp:18 annexb=no) have not convinced the remote endpoint to disable VAD. What makes you think there is a SID with the wrong length, rather than no SID? Do the first 2 of the 10 bytes look like SID? I expect if you have annexb set to no, then some other form of VAD is active, and suppressing transmission. 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] Is it possible to keep both call legs live after Dial()
On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan barry+asterisk-us...@opensolutions.ie wrote: Now, ideally, I would be able to act on a 'decision' from a DTMF sequence from the agent's handset. I don't think this is possible unfortunately. Please correct me if I'm wrong. DYNAMIC_FEATURES within features.conf -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] SMS and fixed land lines
Le 06/09/2010 15:10, Olivier a écrit : Hi, Hello 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? As stated by Philipp, SMSC is unique. However -in France at least- SMS sended to landlines are altered and sended as voice messages by the operators. For messages from Orange you will recognize that's a SMS as the callerID is the Orange SMSCs one. For SFR no luck, Bouygues don't tested. -- Daniel -- _ -- 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] Dial timeout and SIP 302 Moved Temporarily
Hi, With a 1.4.35 or 1.6.1.19, I'm facing this behaviour : - extension 7002 is a SIP hard phone currently configured to forward incoming calls to extension 7003, when a call is unanswered within a 10s time frame - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) statement and no one answers, then : - after 10s, Asterisk receives SIP 302 Moved temporarily message and enters its dialplan to call 7003, as required, - 10s later (or 20s from the very start), call from 7001 to 7003 is cut and the next statement after Dial(SIP/7002,20) is run. The behaviour I would ideally implement is : - whenever a SIP 302 Moved temporarily message is received, timer associated to the original call (the one from 7001 to 7002) is reset to another 20s period Alternatively, I would also to have the first call timer cancelled. At the moment, I think I would try the following : - before or within the Dial(SIP/7002,20), set an inherited variable with the value of the channel to kill is case the call is forwarded, - when dialplan is (re-)entered check is the call is a forwarded one, - if positive, then soft hangup the second leg of the original call, hoping that this would not introduce undesirable side effects. Do you have any suggestion ? 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] Going to go out on a limb here - regarding Vonage
Okay; So I can use a Digium FXO/FXS type card and use the dial tone to utilize Vonage with Asterisk. Done it - simple enough. However.I am wondering if anyone is Cracker-Jack enough to come up with a way to get SIP credentials? I went as far as asking Vonage directly and the answer I got was a big fat NO. I am thinking it is probably a violation of their acceptable use policy to do it - honestly, I have been with Vonage for about 8 years and never read it. I know there are other services out there that will give you the SIP info (Broadvoice). Anyone been successful and willing to share the knowledge? Cheers Glen -- _ -- 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] SMS and fixed land lines
2010/9/6 Administrator TOOTAI ad...@tootai.net Le 06/09/2010 15:10, Olivier a écrit : Hi, Hello 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? For SFR no luck, What do you mean by that ? That SMS from cellphones cannot reach landlines or are not using a unique SMSC callerid which makes them unrecognizable ? Bouygues don't tested. -- Daniel -- _ -- 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] Possible malformed G729B - SID (VAD/DTX)framesfrom carrier endpoint ?
Steve- We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. We have Kamailio + rtpproxy running between the endpoints. Do you think it's reasonable to convert the first malformed SID frame (10 bytes) to 2 bytes, and then strip the following malformed SID frames until we see the talkspurt marker bit is set? We could do that... I'm wondering if anyone has seen such malformed SID frames before. As a couple of additional notes, between us and the remote endpoint there appears to be using an ALOE Systems (formerly MERA systems) MSiP system. So far the SDP negotiations we've tried (e.g. a=fmtp:18 annexb=no) have not convinced the remote endpoint to disable VAD. What makes you think there is a SID with the wrong length, rather than no SID? Do the first 2 of the 10 bytes look like SID? The first two bytes appear not to be a proper SID. However, as Vikram mentioned time-stamps show an increase greater than ptime and MARK bit is set in the RTP header. Then there are several consecutive packets (from 10 to 100) with this combination. Once we see the first of these, possibly we could strip and generate a correct SID. I expect if you have annexb set to no, then some other form of VAD is active, and suppressing transmission. Yes... something in the middle... possibly the MSiP. -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
Re: [asterisk-users] SMS and fixed land lines
Hi! Yes, typically there is only one SMSC that can send you SMS on a fixed line; look at its Caller ID to identify a SMS call. Even when the call is coming from a cellphone ? A SMS is not really a call (at least not in the mobile world), and the cellphone cannot directly send a SMS to a landline phone. Instead it hands the SMS to the SMSC of the mobile carrier, which in turn hands it over to the SMSC of the landline carrier. Philipp -- _ -- 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] SMS and fixed land lines
Le 06/09/2010 17:39, Olivier a écrit : 2010/9/6 Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net Le 06/09/2010 15:10, Olivier a écrit : Hi, Hello 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? For SFR no luck, What do you mean by that ? That SMS from cellphones cannot reach landlines or are not using a unique SMSC callerid which makes them unrecognizable ? No unique SMSC. In the voice message they send you, it's You receive an SMS from John Doe, press 1 if you want to listen the message Very funny when you have your voicemail activated or fax detection before voice :-( The callerID is the one from the SMS sender but this means nothing as you can send SMSs from a ... landline! They are so stupid ... -- Daniel -- _ -- 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 stops processing calls...
I have a very difficult to diagnose problem. We are running Asterisk 1.6.2.11, DAHDI 2.4.0, FreePBX 2.8 on a Centos 5.5 server (Xeon quad core 4gb). Last week we started having a problem where the server will randomly stop sending and receiving calls. Asterisk does not die or crash. You can get the CLI but any command you input will not respond. All phones have No Service on their screens and if you dial into the server you can see the channel event but it never answers. Once we restart Asterisk everything goes back to normal. This is now happening several times a day so obviously the client is pissed. This customer has 4 Asterisk servers which all but this one works well. One of the others is running in the same hardware and environment but does not have this problem. In the log files the only weird thing I see is: [Sep 6 11:09:48] DEBUG[24238] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 49 [Sep 6 11:09:48] DEBUG[24239] audiohook.c: Read factory 0x2c5d3c60 was pretty quick last time, waiting for them. [Sep 6 11:09:48] DEBUG[24288] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 54 [Sep 6 11:09:48] DEBUG[24452] audiohook.c: Write factory 0x2aaacc67ad08 was pretty quick last time, waiting for them. [Sep 6 11:09:48] DEBUG[24492] audiohook.c: Failed to get 160 samples from write factory 0x2aaac8aa2ba8 [Sep 6 11:09:48] DEBUG[24492] audiohook.c: Read factory 0x2aaac8aa2170 and write factory 0x2aaac8aa2ba8 both fail to provide 160 samples These messages are repeated hundreds of times per minute. The only reference I can find to these messages were from Asterisk 1.4.X where recording playback sounded too fast but this is not the case here since recording play at normal speed (plus we are suing 1.6). Any tips on how to properly debug this situation? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] How are shared variables destroyed ?
Hi, How are shared variables destroyed (the one set with function SHARED) ? Shall I care about that or are those variables destroyed whenever associated channel is destroyed ? 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
Re: [asterisk-users] SMS and fixed land lines
On Mon, Sep 6, 2010 at 5:24 PM, Administrator TOOTAI ad...@tootai.net wrote: As stated by Philipp, SMSC is unique. However -in France at least- SMS sended to landlines are altered and sended as voice messages by the operators. For messages from Orange you will recognize that's a SMS as the callerID is the Orange SMSCs one. For SFR no luck, Bouygues don't tested. Actually, in France, if the landline has the extra billed SMS service, the SMS is sent as described by others. There is an extra digit at the end for a kind of mailbox. This dates from when some phones had multiple inboxes for SMS. I used that digit to send difference command codes to my asterisk box, such as call me back, etc. I think if the mailbox was 0, the message was read, or perhaps if you didn't subscribe the line to SMS it was the case. Some of this may have changed, but when I has asterks and a fixed-line SMS service from France Télécom, that's the way it worked. /r -- _ -- 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 Fax
On 09/06/2010 07:45 AM, Andrew Nowrot wrote: Hi I know that this topic was on the list maybe dozen of times. But I have a question regarding the fax support in asterisk, because all the information I could get does not give me the clear view of if. I read that Asterisk 1.8 will have strong fax (t.38) support, but I want to know if these four scenarios will be possible to achieve: fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- SPA2102 ATA --- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- PSTN --- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- IAX --- another Asterisk 1.8 --- SPA2102 ATA--- fax machine fax machine (phone+fax) connected to ATA --- SPA2102 ATA --- Asterisk 1.8 --- SIP --- another Asterisk 1.8 --- SPA2102 ATA --- fax machine For last three scenarios Asterisk should work as fax T.38 gateway. Is it possible? There is no support for T.38 gateway mode in Asterisk 1.8, although there is still work on that front. The patches in the issue tracker may have been updated for Asterisk 1.8 already, though. -- 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] SMS and fixed land lines
Le 06/09/2010 19:31, Randy R a écrit : [...] Some of this may have changed, but when I has asterks and a fixed-line SMS service from France Télécom, that's the way it worked. End of 2009 SMS sended to landlines where easy to treat, we even setup an SMS2Mail gw. Those days, we only treat SMSs from Orange/France Telecom as they SMSC has is own callerID. -- Daniel -- _ -- 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] What can make G.729a codec hostid change?
After upgrading my small test system from Debian Etch-Lenny via a complete reinstall, I find my g729 hostid has changed. Same machine, same CPU, same NIC! It doesn't seem reasonable that I have to burn my one no-hassle re-registration for a simple OS upgrade. The README only says that hostid is based on MAC addresses of all NICs, but that doesn't seem to be true. Does anyone know anything else that might cause g729 to compute a different hostid? Console output follows: Connected to Asterisk SVN-branch-1.6.2-r284958M currently running on secundus (pid = 2430) secundus*CLI g729 show version Digium G.729A Module Version 1.6.2.0_3.1.4 (optimized for k6_3_32) secundus*CLI g729 show hostid Host-ID: 02:e1:6c:f6:81:a7:06:b6:4d:fc:94:49:83:c5:3e:71:a4:0f:1b:2c secundus*CLI g729 show licenses 0/0 encoders/decoders of 0 licensed channels are currently in use Licenses Found: File: G729-2028.lic -- Key: G729-2028 -- Host-ID: 98:3e:89:19:af:0c:11:32:49:cc:fc:9b:e4:92:63:bb:fc:0b:26:4d -- Channels: 0 (incorrect host-id) File: G729-4075.lic -- Key: G729-4075 -- Host-ID: 98:3e:89:19:af:0c:11:32:49:cc:fc:9b:e4:92:63:bb:fc:0b:26:4d -- Channels: 0 (incorrect host-id) -- Barry -- _ -- 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] MeetMe errorhandling
Hi Group,
i have a MeetMe Question.
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference Room
doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the
Call.
there is a solution for the kind my problem?
Thanx and bye
Daniel
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Re: [asterisk-users] MeetMe errorhandling
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference
Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk
Hangup the Call.
Use the i extension to control what happens when entering an invalid room
number. Simple example:
exten = 5000,Goto(confline,s,1)
[confline]
exten = s,1,Background(enter-conf-call-number)
exten = s,n,WaitExten(20)
exten = i,1,Playback(conf-invalid)
exten = i,n,Goto(s,1)
exten = t,1,Goto(s,1)
; Participants always dial a 7-digit conference number, optionally followed
; by the #-sign
exten = _XXX,1,MeetMe(${EXTEN},Mxwsp)
exten = _XXX,n,Hangup()
exten = _XXX#,1,Goto(${EXTEN:-8:7},1)
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Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Tilghman Lesher tles...@digium.com wrote: On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote: Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. That doesn't make any sense. Revision 282911 is a merge to a team branch, nothing related to the 1.8 branch. Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? That was the one next in the logs, maybe I will try latest and see if it goes away. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] asterisk-1.8 problem with one-way audio with no nat
Tilghman Lesher tles...@digium.com wrote: On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote: Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. That doesn't make any sense. Revision 282911 is a merge to a team branch, nothing related to the 1.8 branch. Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? Or to put it another way the last good install for me is 281875 so it right after that where from express talk to an outside line through asterisk is failing with one way audio after the first several seconds. I did try latest update and it is still failing. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] Is it possible to keep both call legs live after Dial()
Dial with M option On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan barry+asterisk-us...@opensolutions.ie wrote: Hi folks, After a fairly extensive Google trawl, I don't think the following is possible but would appreciate confirmation from anyone else who has tried something similar. I have an AGI (not particularly relevant) which is executed when someone calls into a specific extension. This AGI finds a suitable 'agent' (not actually a queuing system in the Asterisk Queue sense) and Dial()s this agent bridging the call. Now, ideally, I would be able to act on a 'decision' from a DTMF sequence from the agent's handset. I don't think this is possible unfortunately. Please correct me if I'm wrong. I can get a 'decision' from the agent by using the 't' Dial() option and have the agent key an extension corresponding to a 'decision'. This will suffice. From this I can call another AGI for the caller and continue processing them. I'd like to be able to play some audio to the agent and even let the agent call continue with another AGI. This bit I don't think is possible either? Thanks and kind regards, Barry -- _ -- 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] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6
Hi Danny, I don't think this is the issue as I get the same problem when I divert one of my SIP handsets to that extension, and dial internally. The connection happens instantly. I can see the file playing on the asterisk console whilst I am getting dead air. aF On 01/09/2010, at 7:54 AM, Danny Nicholas wrote: You're probably not going to buy this, but if custom/ceh-meetingmsg is less than 7 seconds long, it could be playing before the connection is established. -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi Paul, No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw with no success. The ISDN interface is alaw and the SIP phones I was testing with are definately alaw. Not sure what to do from here. I might just need to bypass the issue using some alternate way to put the message in front of the inbound dialplan logic on some condition. aF On 01/09/2010, at 8:06 AM, Paul Belanger wrote: On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara a...@receptiveit.com.au wrote: Hi Paul, I tried adding Progress() to no avail. I still get no audio and below is what comes up in the console. Try moving Progress() before the Dial(). If you Answer() the channel, do you have the same problem? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
