[asterisk-users] Missing voicemail prompt beginning
Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like number 12345 not available I was only hearing 345 not available. Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started correctly in every test case. Any hints on how I can debug this? I think it's some problem on my local configuration, I doubt it's a problem with my SIP provider or mobile phone provider, they are both very reliable (Sipgate and T-Mobile). Thanks for any hint! Best regards Stefan -- _ -- 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] Missing voicemail prompt beginning
Stefan at WPF wrote:
beginning of the prompt was missing
User answer(500) or wait(1) before the audio prompts.
Example:
exten = s,1,Answer(500)
exten = s,n,Voicemail({$ARG1}@sip,u)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
_
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Re: [asterisk-users] Missing voicemail prompt beginning
First of all, thank you for your reply, however I see two problems with
this solution:
1) I think sometimes even more than a second from the beginning of the
prompt is missing, so I have to set a larger value, meaning in cases where
nothing of the prompt was missing, the calling person listens to a pause of
some seconds.
2) Your solutions handles the symptoms of the problem, I'd like to fix the
root cause of this problem.
Any ideas on number 2, fixing / finding the root cause of this problem?
Thanks :-)
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
beginning of the prompt was missing
User answer(500) or wait(1) before the audio prompts.
Example:
exten = s,1,Answer(500)
exten = s,n,Voicemail({$ARG1}@sip,u)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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--
_
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Re: [asterisk-users] Missing voicemail prompt beginning
Stefan at WPF wrote: 2) Your solutions handles the symptoms of the problem, I'd like to fix the root cause of this problem. The root cause of the problem (Most likely) is that the channel hadn't be answered. A wait, allows the channel to be established and audio to pass. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Missing voicemail prompt beginning
2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: 2) Your solutions handles the symptoms of the problem, I'd like to fix the root cause of this problem. The root cause of the problem (Most likely) is that the channel hadn't be answered. A wait, allows the channel to be established and audio to pass. Which end do you mean with channel not answered? The asterisk end or mobile phone end of the channel? Also I am confused that it sometimes work and sometimes it does not? ): I tried with Wait(1) and Answer(1000), unfortunately both didn't change things - sometimes the complete prompt is there, sometimes the beginning is missing ): Are there any relevant logs for these things / how to check what the problem is without trying settings? Thanks :-) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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] Missing voicemail prompt beginning
Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Missing voicemail prompt beginning
Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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] Missing voicemail prompt beginning
Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- _ -- 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] Missing voicemail prompt beginning
Thank you Warren, I will temporarily skip this step, as I don't have the problem anymore, though I don't know why (for that and learning purposes the logs maybe would be still useful). I found some different settings for Asterisk and Sipgate (actually I found the settings for private users on the Sipgate website, before that I found the settings for business customers and assumed there wouldn't be a difference). When I had the problem, my sip.conf looked like this: [general] port=5060 bindaddr=0.0.0.0 context=other language=de register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=peer context=from_external_voip_provider username=SIPID defaultuser=SIPID fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes Now my sip.conf looks like this (source: http://www.sipgate.de/faq/index.php?do=displayArticlearticle=540id=257): (I have commented the additions / changes) [general] port=5060 bindaddr=0.0.0.0 context=other language=de qualify=no ; added disallow=all ; added allow=alaw ; added allow=ulaw ; added allow=g729 ; added allow=gsm; added allow=slinear; added srvlookup=yes; added register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=friend ; changed from peer to friend context = from_external_voip_provider username=SIPID ;defaultuser=SIPID ; removed fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes canreinvite=no ;added dtmfmode=rfc2833 ;added The dialplan in both cases was this: [from_external_voip_provider] exten = SIPID,1,Answer(1000) exten = SIPID,n,VoiceMail(some_number,u) exten = SIPID,n,Hangup() (I left out the Dial command for testing purposes after I found the voicemail prompt problems) If anyone has an idea why it now works without problems, please let me know for learning purposes. I still have to read up on the options. When I have more time I will probably also set the old settings again to learn how I could have identified the problem. 2012/6/17 Warren Selby wcse...@selbytech.com Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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 -- _ -- 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:
Re: [asterisk-users] Missing voicemail prompt beginning
Sorry for the second mail, about the infrastructure: phone - asterisk - HW firewall including NAT - Sipgate SIP Provider About Software: Asterisk 1.8.13.0 running on Raspbian Debian Linux (http://www.raspbian.org/, Raspbian includes up to date Asterisk paackages while the normal Raspberry Pi Debian does not) running on a Raspberry Pi http://www.raspberrypi.org/:-) 2012/6/17 Stefan at WPF stefan.at@googlemail.com Thank you Warren, I will temporarily skip this step, as I don't have the problem anymore, though I don't know why (for that and learning purposes the logs maybe would be still useful). I found some different settings for Asterisk and Sipgate (actually I found the settings for private users on the Sipgate website, before that I found the settings for business customers and assumed there wouldn't be a difference). When I had the problem, my sip.conf looked like this: [general] port=5060 bindaddr=0.0.0.0 context=other language=de register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=peer context=from_external_voip_provider username=SIPID defaultuser=SIPID fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes Now my sip.conf looks like this (source: http://www.sipgate.de/faq/index.php?do=displayArticlearticle=540id=257): (I have commented the additions / changes) [general] port=5060 bindaddr=0.0.0.0 context=other language=de qualify=no ; added disallow=all ; added allow=alaw ; added allow=ulaw ; added allow=g729 ; added allow=gsm; added allow=slinear; added srvlookup=yes; added register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=friend ; changed from peer to friend context = from_external_voip_provider username=SIPID ;defaultuser=SIPID ; removed fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes canreinvite=no ;added dtmfmode=rfc2833 ;added The dialplan in both cases was this: [from_external_voip_provider] exten = SIPID,1,Answer(1000) exten = SIPID,n,VoiceMail(some_number,u) exten = SIPID,n,Hangup() (I left out the Dial command for testing purposes after I found the voicemail prompt problems) If anyone has an idea why it now works without problems, please let me know for learning purposes. I still have to read up on the options. When I have more time I will probably also set the old settings again to learn how I could have identified the problem. 2012/6/17 Warren Selby wcse...@selbytech.com Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__** _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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 -- _ -- 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
