Re: [asterisk-users] Can't make call with TDM410P
On Saturday 23 June 2012, neo haux wrote: Actually I can start and receive SIP calls (PC client, iphone client) but I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I notice the number you Dial()led didn't start with a zero. Check with your telco about this if you like but you almost certainly need to include the initial 0 of the STD code when using an analogue exchange line, because a TDM410P simply emulates standard subscriber's apparatus. Basically, your Asterisk box is just a subscriber dialling out on a POTS phone; and it has to do exactly what a person with a cheap phone would do. That is, the STD code (including initial 0) and subscriber's number for someone in a different town; or just the number for a call to someone in the same town (though dialling the code for a local call won't break anything). For IDD you need to dial 00, wait awhile, then the code for the destination country, STD code *without* the initial 0 (note, some small countries don't use STD codes) then subscriber's number. -- AJS Answers come *after* questions. -- _ -- 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] Proactive problem monitoring on SIP on Asterisk
Le 21/06/2012 09:52, Ishfaq Malik a écrit : On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote: Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. [...] I've not used this myself but had a look at the site and I think it's pretty much what you're after... http://www.voipmonitor.org/ It works well, people are reactive -- 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] FastAGI script and DIAL execution
Hi all, I am trying to control the whole call using a FastAGI script. To that effect I launch a FastAGI script (written with asterisk-java). Basically, I want to DIAL from within the FastAGI script. When the call ends I want to control the hangup (if executed at the remote end), and depending on the cause, dial again, play a message, or hang up. This is a pretty standard telephony scenario. I did it before by executing the AGI, setting variables, calling the DIAL command from the dialplan, and then executing a second AGI script for the cleanup logic. However, now that I am using FastAGI it seems like a better idea to keep the AGI script alive during the duration of the call. This gives me a lot of control and fexibility on reporting. However, as far as I can tell, once the called party hangs up, the CDR is generated and posted, _even though my script is still in execution_! As you can see from the sample below, the called party hangs up, and dialplan execution starts immediately at the h extension, even though my script is still running. In fact, I have quite a bit of cleanup to do, adding variables to the CDR's, and none of them are saved! I believe this is because the CDR is already finised. It's like if once you call the DIAL aplication, the dialplan forks off and your script is running in a different place. I do not understand it. I assumed when I called DIAL from within a script, that the script execution would suspend, but be resumed once the DIAL command returned, but this is not what is happening. Is there any way to get that behaviour? Regards, Alex Entering customer extension -- Executing [62999@customer:2] Verbose(SIP/139255423-004c, 5,Dialed - 62999) in new stack Dialed - 62999 -- Executing [62999@customer:3] Set(SIP/139255423-004c, origincontext=customer) in new stack -- Executing [62999@customer:4] Goto(SIP/139255423-004c, transform,62999,1) in new stack -- Goto (transform,62999,1) -- Executing [62999@transform:1] Goto(SIP/139255423-004c, customer,003462999,transform) in new stack -- Goto (customer,003462999,5) -- Executing [003462999@customer:5] Verbose(SIP/139255423-004c, 5,New dialnum - 003462999) in new stack New dialnum - 003462999 -- Executing [003462999@customer:6] Set(SIP/139255423-004c, CDR(server)=7) in new stack -- Executing [003462999@customer:7] Set(SIP/139255423-004c, CDR(srcip)=) in new stack -- Executing [003462999@customer:8] AGI(SIP/139255423-004c, agi://localhost/auth) in new stack AGI Tx agi_network: yes AGI Tx agi_network_script: auth SIP/139255423-004cAGI Tx agi_request: agi://localhost/auth SIP/139255423-004cAGI Tx agi_channel: SIP/139255423-004c SIP/139255423-004cAGI Tx agi_language: es SIP/139255423-004cAGI Tx agi_type: SIP SIP/139255423-004cAGI Tx agi_uniqueid: 1340616655.76 SIP/139255423-004cAGI Tx agi_version: 10.5.0 SIP/139255423-004cAGI Tx agi_callerid: 139255 SIP/139255423-004cAGI Tx agi_calleridname: unknown SIP/139255423-004cAGI Tx agi_callingpres: 0 SIP/139255423-004cAGI Tx agi_callingani2: 0 SIP/139255423-004cAGI Tx agi_callington: 0 SIP/139255423-004cAGI Tx agi_callingtns: 0 SIP/139255423-004cAGI Tx agi_dnid: 62999 SIP/139255423-004cAGI Tx agi_rdnis: unknown SIP/139255423-004cAGI Tx agi_context: customer SIP/139255423-004cAGI Tx agi_extension: 003462999 SIP/139255423-004cAGI Tx agi_priority: 8 SIP/139255423-004cAGI Tx agi_enhanced: 0.0 SIP/139255423-004cAGI Tx agi_accountcode: 704741 SIP/139255423-004cAGI Tx agi_threadid: 1104279872 SIP/139255423-004cAGI Tx SIP/139255423-004cAGI Rx GET VARIABLE CDR(src) SIP/139255423-004cAGI Tx 200 result=1 (139255423) SIP/139255423-004cAGI Rx SET VARIABLE CDR(accountcode) 704741 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(dest_id) 507 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(routeplan) 11261 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(carrier) 69 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx EXEC Dial SIP/10003462999@x.x.x.x -- AGI Script Executing Application: (Dial) Options: (SIP/10003462999@x.x.x.x) == Using SIP RTP CoS mark 5 -- Called SIP/10003462999@193.17.66.71 -- SIP/193.17.66.71-004d is making progress passing it to SIP/139255423-004c -- SIP/193.17.66.71-004d is ringing -- SIP/193.17.66.71-004d is making progress passing it to SIP/139255423-004c -- SIP/193.17.66.71-004d answered SIP/139255423-004c -- Executing [h@customer:1] Set(SIP/139255423-004c, CDR(q931)=16) in new stack -- Executing [h@customer:2] Set(SIP/139255423-004c,
Re: [asterisk-users] IAX Trunk issue.
On 06/24/2012 07:53 PM, Mitchell Johnson wrote:
I'm testing a few IAX trunk scenarios in a controlled lab. From server2
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across
the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the
6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also
plays tt-weasels as it's supposed to, but it's not the tt-weasels under its
extension. It also dials the s extension.
I only placed the s extension in the dial plan to verify that the traffic was
going across the IAX trunk and hitting the correct context.
Any help would be greatly appreciated.
Thanks Mitch
[phones]
exten = _60XX,1,Dial(IAX2/trunk-1)
exten = _X.,1,Dial(IAX2/trunk-1)
exten = 5000,1,Dial(SIP/${EXTEN})
exten = 5000,n,Hangup
same = n,Hangup()
exten = 5099,1,Playback(tt-monkeys)
exten = 5099,n,HangUp
You are not telling asterisk-1 where you want the call to go, so it is going to
's'.
Try adding the extension to the Dial() command on asterisk-2. Change
Dial(IAX2/trunk-1)
to
Dial(IAX2/trunk-1/${EXTEN})
Note: It appears that you are doing it correctly from asterisk-1
towards asterisk-2
exten = _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
Assuming, of course, that the variable IAXTrunk is properly set.
Dale
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
--
_
-- 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] Voicemail attachment format
Hi All, I have a simple urgent question that I couldn't find the answer yet, can we customize the voicemail attachment format *per user* in asterisk *1.2 *(like all receive wav attch but one or two users receive attch in gsm format)? if yes can you show me how please? -- Khalid Touati -- _ -- 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 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP
On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote: Have a look at the latest blacklist sample in dahdi trunk http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log file: blacklist.sample ... # Some mISDN drivers may try to attach to cards supported by DAHDI. If you # have a card which is *not* supported by DAHDI but supported by one of the # below drivers you should feel free to remove it from the blacklist below. blacklist hfcmulti May collide with wcb4xxp blacklist netjet May collide with wctdm and some other older drivers. blacklist hfcpci May collide with zaphfc. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP
2012/6/25, Tzafrir Cohen tzafrir.co...@xorcom.com: On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote: Have a look at the latest blacklist sample in dahdi trunk http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log file: blacklist.sample ... # Some mISDN drivers may try to attach to cards supported by DAHDI. If you # have a card which is *not* supported by DAHDI but supported by one of the # below drivers you should feel free to remove it from the blacklist below. blacklist hfcmulti May collide with wcb4xxp blacklist netjet May collide with wctdm and some other older drivers. blacklist hfcpci May collide with zaphfc. May I ask where I can get this zaphfc from ? Is this the same refered to as vzaphfc (see http://www.voip-info.org/wiki/view/Asterisk+vzaphfc) ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Digium IP Phones D40
On 06/22/2012 05:12 PM, bilal ghayyad wrote: One of the problems I faced with Polycom is the voice volume and ring volume, it is low. When it rings, even if it is maximum volume, still it is weak. When I talk and I set the volume to the maximum, I still feel the voice volume is low and would if to increase it. I have never, in over 7 years of using Polycom phones, heard anyone complain that the maximum volume was too low. Most devices of this type have their maximum volume controlled to meet guidelines set by government and industry recommendations (in order to avoid causing damage to users' ears), and the Digium phones are no exception. If you are finding that the volume produced by common SIP phones is too low and you can't make it loud enough, I'd bet that the problem is not in the phones, but in your environment or your ears :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Voicemail attachment format
On Mon, Jun 25, 2012 at 9:23 AM, khalid touati khalidtou...@gmail.comwrote: Hi All, I have a simple urgent question that I couldn't find the answer yet, can we customize the voicemail attachment format *per user* in asterisk *1.2 *(like all receive wav attch but one or two users receive attch in gsm format)? if yes can you show me how please? I don't think that was an option in 1.2, but I haven't used 1.2 in so long I may be off. Hopefully one of our resident 1.2 luddite's will see this and have a more definitive answer for you. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com 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] Digium IP Phones D40
On Mon, 2012-06-25 at 10:45 -0500, Kevin P. Fleming wrote: On 06/22/2012 05:12 PM, bilal ghayyad wrote: One of the problems I faced with Polycom is the voice volume and ring volume, it is low. When it rings, even if it is maximum volume, still it is weak. When I talk and I set the volume to the maximum, I still feel the voice volume is low and would if to increase it. I have never, in over 7 years of using Polycom phones, heard anyone complain that the maximum volume was too low. Most devices of this type have their maximum volume controlled to meet guidelines set by government and industry recommendations (in order to avoid causing damage to users' ears), and the Digium phones are no exception. If you are finding that the volume produced by common SIP phones is too low and you can't make it loud enough, I'd bet that the problem is not in the phones, but in your environment or your ears :-) Actually we get that complaint a lot too (Polycom ring volume). We typically install in hotel environments, and in their back office the environment can be noisy, as well as in their restaurants. I imagine in a typical office environment this wouldn't be an issue... 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
Re: [asterisk-users] Digium IP Phones D40
On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote: Actually we get that complaint a lot too (Polycom ring volume). We typically install in hotel environments, and in their back office the environment can be noisy, as well as in their restaurants. I imagine in a typical office environment this wouldn't be an issue... The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The newer SPA5xx series have been seriously toned down. I'm guessing it's all 'safety' related :S 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] Digium IP Phones D40
We have the ringer volume issue with some customer environments as well. We use Grandstream phones in a lot of installs so we just upload a custom ringtone with the db pushed up on it a bit. We are testing the Digium phones and have concerns if we will be able to use them for the high noise env customers. Polycom phones do have the same ring volume issue for these customers. No issues in general office env. Bryant From: Steven Howes steve-li...@geekinter.net Sent: Monday, June 25, 2012 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium IP Phones D40 On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote: Actually we get that complaint a lot too (Polycom ring volume). We typically install in hotel environments, and in their back office the environment can be noisy, as well as in their restaurants. I imagine in a typical office environment this wouldn't be an issue... The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The newer SPA5xx series have been seriously toned down. I'm guessing it's all 'safety' related :S 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
Re: [asterisk-users] Digium IP Phones D40
- Original Message - We have the ringer volume issue with some customer environments as well. We use Grandstream phones in a lot of installs so we just upload a custom ringtone with the db pushed up on it a bit. We are testing the Digium phones and have concerns if we will be able to use them for the high noise env customers. Polycom phones do have the same ring volume issue for these customers. No issues in general office env. For everyone complaining about the Polycom's lack of volume, are you simply hitting the volume buttons, or are you also aware of the myriad of adjustments available in the Polycom XML provisioning configs? We have a local educational customer that experienced volume problems with Polycom due to noisy classroom environments, and with a few tweaks to volume and gain in the XML configs pushed via TFTP, the phones were ear-splittingly loud, both ringers and handset. --Tim -- _ -- 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 options
I am looking for a CDR report tool that will link extensions to the user's names... are there any that offer this feature? We are using trixbox 2.8. -- _ -- 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] SendFAX timestamp
Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] FastAGI script and DIAL execution
Alejandro, Try the 'g' option to Dial(): http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial - *g*: When the called party hangs up, continue to execute commands in the current context at the next priority On 25 June 2012 20:17, Alejandro Recarey alexreca...@gmail.com wrote: Hi all, I am trying to control the whole call using a FastAGI script. To that effect I launch a FastAGI script (written with asterisk-java). Basically, I want to DIAL from within the FastAGI script. When the call ends I want to control the hangup (if executed at the remote end), and depending on the cause, dial again, play a message, or hang up. This is a pretty standard telephony scenario. I did it before by executing the AGI, setting variables, calling the DIAL command from the dialplan, and then executing a second AGI script for the cleanup logic. However, now that I am using FastAGI it seems like a better idea to keep the AGI script alive during the duration of the call. This gives me a lot of control and fexibility on reporting. However, as far as I can tell, once the called party hangs up, the CDR is generated and posted, _even though my script is still in execution_! As you can see from the sample below, the called party hangs up, and dialplan execution starts immediately at the h extension, even though my script is still running. In fact, I have quite a bit of cleanup to do, adding variables to the CDR's, and none of them are saved! I believe this is because the CDR is already finised. It's like if once you call the DIAL aplication, the dialplan forks off and your script is running in a different place. I do not understand it. I assumed when I called DIAL from within a script, that the script execution would suspend, but be resumed once the DIAL command returned, but this is not what is happening. Is there any way to get that behaviour? Regards, Alex Entering customer extension -- Executing [62999@customer:2] Verbose(SIP/139255423-004c, 5,Dialed - 62999) in new stack Dialed - 62999 -- Executing [62999@customer:3] Set(SIP/139255423-004c, origincontext=customer) in new stack -- Executing [62999@customer:4] Goto(SIP/139255423-004c, transform,62999,1) in new stack -- Goto (transform,62999,1) -- Executing [62999@transform:1] Goto(SIP/139255423-004c, customer,003462999,transform) in new stack -- Goto (customer,003462999,5) -- Executing [003462999@customer:5] Verbose(SIP/139255423-004c, 5,New dialnum - 003462999) in new stack New dialnum - 003462999 -- Executing [003462999@customer:6] Set(SIP/139255423-004c, CDR(server)=7) in new stack -- Executing [003462999@customer:7] Set(SIP/139255423-004c, CDR(srcip)=) in new stack -- Executing [003462999@customer:8] AGI(SIP/139255423-004c, agi://localhost/auth) in new stack AGI Tx agi_network: yes AGI Tx agi_network_script: auth SIP/139255423-004cAGI Tx agi_request: agi://localhost/auth SIP/139255423-004cAGI Tx agi_channel: SIP/139255423-004c SIP/139255423-004cAGI Tx agi_language: es SIP/139255423-004cAGI Tx agi_type: SIP SIP/139255423-004cAGI Tx agi_uniqueid: 1340616655.76 SIP/139255423-004cAGI Tx agi_version: 10.5.0 SIP/139255423-004cAGI Tx agi_callerid: 139255 SIP/139255423-004cAGI Tx agi_calleridname: unknown SIP/139255423-004cAGI Tx agi_callingpres: 0 SIP/139255423-004cAGI Tx agi_callingani2: 0 SIP/139255423-004cAGI Tx agi_callington: 0 SIP/139255423-004cAGI Tx agi_callingtns: 0 SIP/139255423-004cAGI Tx agi_dnid: 62999 SIP/139255423-004cAGI Tx agi_rdnis: unknown SIP/139255423-004cAGI Tx agi_context: customer SIP/139255423-004cAGI Tx agi_extension: 003462999 SIP/139255423-004cAGI Tx agi_priority: 8 SIP/139255423-004cAGI Tx agi_enhanced: 0.0 SIP/139255423-004cAGI Tx agi_accountcode: 704741 SIP/139255423-004cAGI Tx agi_threadid: 1104279872 SIP/139255423-004cAGI Tx SIP/139255423-004cAGI Rx GET VARIABLE CDR(src) SIP/139255423-004cAGI Tx 200 result=1 (139255423) SIP/139255423-004cAGI Rx SET VARIABLE CDR(accountcode) 704741 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(dest_id) 507 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(routeplan) 11261 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx SET VARIABLE CDR(carrier) 69 SIP/139255423-004cAGI Tx 200 result=1 SIP/139255423-004cAGI Rx EXEC Dial SIP/10003462999@x.x.x.x -- AGI Script Executing Application: (Dial) Options: (SIP/10003462999@x.x.x.x) == Using SIP RTP CoS mark 5 -- Called SIP/10003462999@193.17.66.71 -- SIP/193.17.66.71-004d is making progress passing it to SIP/139255423-004c -- SIP/193.17.66.71-004d
Re: [asterisk-users] low success rate for ReceiveFax
In what way was my question not meaningful? Not enough details? Here's our current receive fax route: sender fax machine - telco - E1 line - sangoma card - asterisk We're currently using free fax for asterisk. I have read that fax over voip is not reliable, but is it the same case for faxes going through dahdi channels? It's strange because I previously tested using another asterisk server to send fax using SIP to the receiving server above, and the completion rate is better than using an actual fax machine. From the asterisk console I can see the receiving fax session running, but halfway it stops due to timeout or hangup. Below is a fax session output which was marked as failed: -- Channel 'DAHDI/i1/-4' receiving FAX '/var/spool/asterisk/fax/fax-65126150-1340338724-rx.tif' -- Channel 'DAHDI/i1/-4' FAX session '0' started -- FAX handle 0: [ 000.51 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.98 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000129 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000148 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000174 ], STAT_INFO_DIS Channel 'DAHDI/i1/-4' fax session '0', [ 000.079050 ], channel sent 3 frames (60 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 000.093757 ], stack sent 4 frames (80 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 000.153752 ], stack sent 3 frames (60 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 000.459077 ], channel sent 19 frames (380 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 003.154772 ], stack sent 150 frames (3000 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 003.199289 ], channel sent 137 frames (2740 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 003.211770 ], stack sent 3 frames (60 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 003.259286 ], channel sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.250881 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'DAHDI/i1/-4' fax session '0', [ 005.571646 ], stack sent 118 frames (2360 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 005.599474 ], channel sent 117 frames (2340 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 005.799493 ], channel sent 10 frames (200 ms) of silence. -- FAX handle 0: [ 007.213920 ], STAT_INFO_DCS -- FAX handle 0: [ 007.213946 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 007.213969 ], STAT_NEG_V29_9600 -- FAX handle 0: [ 007.213983 ], STAT_NEG_MMR -- FAX handle 0: [ 007.213995 ], STAT_NEG_A4 -- FAX handle 0: [ 007.214007 ], STAT_NEG_RES_204x98 -- FAX handle 0: [ 007.214019 ], STAT_NEG_ECM -- FAX handle 0: [ 007.214031 ], STAT_EVT_SW_ECM st: WT_DIS_RSP rt: WDSRNSWE Channel 'DAHDI/i1/-4' fax session '0', [ 007.279603 ], channel sent 74 frames (1480 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 007.439604 ], channel sent 8 frames (160 ms) of silence. -- FAX handle 0: [ 007.553962 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'DAHDI/i1/-4' fax session '0', [ 009.219725 ], channel sent 89 frames (1780 ms) of energy. -- FAX handle 0: [ 009.253979 ], STAT_EVT_RX_TRN_END st: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 009.254005 ], STAT_FRM_CFR Channel 'DAHDI/i1/-4' fax session '0', [ 009.414674 ], stack sent 192 frames (3840 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 009.459747 ], channel sent 12 frames (240 ms) of silence. -- FAX handle 0: [ 010.439088 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'DAHDI/i1/-4' fax session '0', [ 010.772670 ], stack sent 68 frames (1360 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 010.799850 ], channel sent 67 frames (1340 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 011.019872 ], channel sent 11 frames (220 ms) of silence. -- FAX handle 0: [ 011.132976 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 011.133002 ], P30EVN_PHASE_C -- FAX handle 0: [ 011.133018 ], P30EVN_DOC_START -- FAX handle 0: [ 011.133049 ], P30EVN_PAGE_START Channel 'DAHDI/i1/-4' fax session '0', [ 014.740131 ], channel sent 186 frames (3720 ms) of energy. -- FAX handle 0: [ 014.812946 ], STAT_EVT_RX_IMG_END st: RCV_ECM rt: RECMNERI Channel 'DAHDI/i1/-4' fax session '0', [ 014.860152 ], channel sent 6 frames (120 ms) of silence. -- FAX handle 0: [ 016.273967 ], STAT_INFO_PPS_EOP -- FAX handle 0: [ 016.273993 ], STAT_EVT_PPS_EOP st: F_END_ECM rt: FEEMNP_P -- FAX handle 0: [ 016.274055 ], P30EVN_PAGE_END -- FAX handle 0: [ 016.274071 ], P30EVN_DOC_END -- FAX handle 0: [ 016.274086 ], STAT_FRM_MCF Channel 'DAHDI/i1/-4' fax session '0', [ 016.340258 ], channel sent 74 frames (1480 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 016.434711 ], stack sent 283 frames (5660 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 016.480280 ], channel sent 7 frames
