[asterisk-users] I'm not able hearing the voice.
Dear Folks, I'm not able hearing the voice of client but on other hand client able to hearing my voice.I'm not able to find out the problem where is i'm wrong. I'm getting continues following error: chan_sip.c:10391 check_via: '' is not a valid host Configuration DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Regards akihlesh -- _ -- 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] Connect to remote GW
On 04/02/14 18:56, Meadows Hoa wrote: If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support both host IP addresses. There is no DNS so straight IP addressing is used. Doing it in the dialplan is a bit of a bodge but perfectly possible. Just make sure you have qualify=yes so if a GW does down asterisk spots it and the Dial() command returns quickly. You might also wish to reduce the check interval down from the default of every minute. The proper way is to make use of DVS SRV however Asterisk doesnt support it properly (doesnt fail over). Not sure if that has been fixed in version 12. If there is just one active GW and one or more backups then the its normally fairly easy to create a virtual IP address on the GW so if one fails then another takes over. Only works if they are on the same network though. Alternatively use something like opensips as a front end. Thats what we do and it load balances between multiple asterisk server and detects if one fails. You can make use of keepalived to provide a virtual IP address which moves between boxes if one dies or the opensips process stops running. -- _ -- 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] Repeated Locally bridging messages
We have a customer reporting poor quality calls when they come to us via a particular provider. The SIP traces look perfectly normal both on the ingress to us and egress to another telco. No additional sip messages after the call has been answered until the BYE is received. However in the asterisk logs I am getting this :- 2014-02-05 13:45:03 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:06 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 Any idea what could be causing this? I am running asterisk 11.2-cert2. I am going to get call redirected via our test box and turn on full verbosity in the logs and capture a full tcpdump but any ideas would be welcome. Thanks Gareth -- _ -- 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] answering machine screening with MixMonitor
I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. I am currently using MixMonitor to append the audio to a named pipe (mkfifo /home/asterisk/var/soundpipe.au), as follows (extensions.conf): [from-pstn] exten = s,1,Wait(20) exten = s,2,Answer exten = s,3,Wait(1) exten = s,4,Set(TIMEOUT(response)=10) exten = s,5,MixMonitor(/home/asterisk/var/soundpipe.au,av(2)) exten = s,6,Background(/home/asterisk/sounds/main) I wrote a small script to loop opening the named pipe and then copying to /usr/bin/play from the sox package. MixMonitor uses the filename extension to determine what format to use for the audio data. I tried all of the formats common to both asterisk (core show file formats) and play (as described on the man page), except for raw. I'm not sure if raw is compatible with sox play - I need to determine bit rates and other parameters before I can try it. I found that in all cases I tested, there was significant latency between the audio on the PSTN line vs. the audio played from the speakers, on the order of 3-10 seconds depending on the format specified. Based on some debug output from my pipe-reader script, it seemed that asterisk opened the write end of the pipe immediately but did not start writing data until the aforementioned delay had passed. au and sln had the lowest latency (3 seconds), so I'm using au for now. Is there any way to reduce the startup latency and make MixMonitor write the audio stream to the output file immediately? I looked briefly at apps/app_mixmonitor.c and main/file.c but I don't fully understand the code. Is mixmonitor forking an external conversion process to generate the audio data? thanks for any insights! -- G. Paul Ziemba FreeBSD unix: 9:06AM up 10 days, 11:05, 4 users, load averages: 1.39, 1.50, 1.54 -- _ -- 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] auto-answer call
thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] answering machine screening with MixMonitor
On 2/5/2014 12:09 PM, G. Paul Ziemba wrote: I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. A local speaker connected to the Asterisk box itself? Console channel driver, chan_alsa (or chan_oss for old drivers). You'll probably end up with kind of a Rube Goldbergish approach, probably something involving ChanSpy or a conferencebridge to take the place of mixmonitor. -- _ -- 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] answering machine screening with MixMonitor
On Wed, 5 Feb 2014 17:09:34 + (UTC) G. Paul Ziemba pz-asterisk-us...@ziemba.us wrote: I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. Have you considered using chan_alsa or chan_oss? I don't know much about it, but I've heard that you can use the sound card in the computer as a phone. If you only want to listen, you wouldn't need a microphone. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] auto-answer call
Please don't top post. On Wed, 5 Feb 2014, Salaheddine Elharit wrote: i test this solution but the issue still the same How does what you see in wireshark compare to what the snom expects? Can you enable debug/verbose syslogging on the phone to see if it complains about anything? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] CDR(start) returns nothing in Asterisk 12
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the
${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
correct values. Where is my mistake? Has this function being renamed?
Leandro
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Re: [asterisk-users] CDR(start) returns nothing in Asterisk 12
On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
the ${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
correct values. Where is my mistake? Has this function being renamed?
This was just fixed yesterday. See
https://issues.asterisk.org/jira/browse/ASTERISK-23250
Richard
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Re: [asterisk-users] Delaying retry since we're currently running
Hi, all I also got the same trouble. If the permission of call file was wrong, asterisk should not add lines after the tail of call files such as DelayRetry . Does anyone help me to solve it? My call file is: = Channel:SIP/192.168.1.200/01124 Callerid: MaxRetries:0 RetryTime:600 WaitTime:60 Context:from-1 Extension:01124 Priority:1 StartRetry: 3284 1 (1391598647) DelayedRetry: 3284 0 (1391598646) DelayedRetry: 3284 0 (1391598647) DelayedRetry: 3284 0 (1391598647) (many the same delayretry information skips) Best regards, Charles 2012-12-28 Danny Nicholas da...@debsinc.com: My best guess is that you are creating the .call file with permissions that don’t allow Asterisk to delete it when it is finished or retries have been exhausted. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir *Sent:* Friday, December 28, 2012 7:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Delaying retry since we're currently running Hi, I am making 200 call concurrently via call files. But i get these messages in asterisk logs: *Delaying retry since we're currently running* Also, in call files i have the following lines: *DelayedRetry: 28662 0 (1356701828)* *DelayedRetry: 28662 0 (1356702128)* *DelayedRetry: 28662 0 (1356702428)* I set MaxRetries: 0. I did not understand the problem, any idea? -- Necati DEMİR -- _ -- 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 -- Best Regards Charles -- _ -- 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] CDR(start) returns nothing in Asterisk 12
I love you all
:-)
Leandro
2014-02-05 Richard Mudgett rmudg...@digium.com:
On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.comwrote:
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
the ${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
correct values. Where is my mistake? Has this function being renamed?
This was just fixed yesterday. See
https://issues.asterisk.org/jira/browse/ASTERISK-23250
Richard
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[asterisk-users] Pedantic=yes not working
Hi List, I have a provider that uses 6060# as a prefix, but when I send the INVITE asterisk is changing the number to 6060%23. I have activated pedantic=yes in the sip.conf but it seems not working at all. I have asterisk 11.7.0. Can someone please guide me here? Thanks, Wilmar -- _ -- 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] auto-answer call
On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, Here is a list of headers used for various vendors, I can't remember which one is for Polycom. SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Alert-Info: Info=Alert-Autoanswer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); SIPAddHeader(Answer-Mode: Auto); Larry. -- _ -- 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] cdr_custom.conf in V12
Hello Asterisk, I just got V12 running and all seems well but just now I looked at my CDR logs and they were messed up so I copied over the sample cdr_custom.conf and uncommented the first master line and the simple line and the logs look like: Simple.csv: 1391652220,, Master.csv ,, And each call creates 6 lines in the file, even if I just call from my cell, answer the phone and then hang up. Either what am I doing wrong or where is cdr_custom.conf documented? I llloked on the wiki but only found documentation for 1.8. -- Ira-- _ -- 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
