[asterisk-users] Sangoma transcoding card bug - drops audio samples
There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have any experience with using Digium cards in such scenario? -- _ -- 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] Call files without permission for asterisk to read
Thanks for the responses. Touching a file after setting permissions does not work. Asterisk only looks at the new file only, not all the files in the directory. Restarting asterisk does work, but dont want to do this. Best way i think would be, as suggested by JG, to sync in a tmp directory and at the time of switch-over mv to outgoing directory. Cheers On Fri, Nov 22, 2013 at 12:36 AM, Steve Edwards wrote: > On Thu, 21 Nov 2013, Rizwan Hisham wrote: > > Hi all,I am syncing call files on my secondary asterisk server but >> without permission to read for asterisk. So they should be executed when I >> grant the right permissions (thats when my primary asterisk server crashes >> or shutsdown somehow). But asterisk only tries to read the file at the time >> of placing the file. So when i grant right permissions nothing happens. Is >> there any workaround to this problem? >> > > When you activate the secondary, 'touch' the files in the spool directory. > > > -- > 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 > -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] SIP FXS ATA with Gigabit ethernet bridge port,
Hi Folks, Is there any SIP FXS ATA with Gigabit ethernet bridge port, in the market ? -- Isamar Maia Cel. VIVO SSA: (55) 71-9940-2012 Cel. TIM SSA: (55) 71-9289-5128 Cel. Claro SSA: (55) 71-9146-8575 Fixo: (55) 71-4062-8688 Skype ID: isamar.maia "A vida é muito curta para ser pequena" (Benjamin Disraeli) -- _ -- 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] Caller's phone keeps ringing after 200 OK
Hello Everyone, I have a strange issue where the caller's phone keeps ringing even after the 200 OK. I am using the latest version of Asterisk 1.8, and wanted to know if anyone could give me any pointers before posting the SIP debug messages. Kind Regards, Nick from Toronto. -- _ -- 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] Movistar sip Mexico
On 21/11/2013 3:32 AM, Damian Gonzalez wrote: Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian Perhaps you could add the following to the peer configuration faxdetect=no 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
Re: [asterisk-users] Movistar sip Mexico
On 22/11/2013 6:49 AM, Alyed wrote: Have you followed the instructions in: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway ?? If possible try with a different ATA since it seems not all of them work fine with fax pass trough. Alyed My understanding of the original posting is that when a voice call arrives from the SIP provider it includes T38 information though the user only wants to accept the g729 component of the call and carry out a voice conversation. If a fax is being received by the SIP provider it only has a the T38 information for the call thus no audio (g729) information is in the SIP message. I don't believe the original poster is attempting to receive a facsimile, instead use voice calls. 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
Re: [asterisk-users] Movistar sip Mexico
Why? On Wednesday, 20 November 2013, Damian Gonzalez wrote: > Hello, > > I have a problem with movistar in Mexico with a sip calls. Movistar send > to me T38 and G729 in the INVITE and they say that I have to ignore T38 and > use G729 in the voice call. > > When a fax call is made Movistar send only T38 in the INVITE. > > Invite example: > > v=0 > o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 > s=sip call > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 6370 RTP/AVP 18 101 > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > m=image 6372 udptl t38 > a=T38FaxVersion:0 > a=T38FaxMaxBuffer:1100 > a=T38FaxMaxDatagram:612 > a=T38MaxBitRate:14400 > a=T38FaxRateManagement:transferredTCF > a=T38FaxUdpEC:t38UDPRedundancy > > How can I ignore T38 and use only G729 for this call?. > > Thanks for your help. > > Damian > > > -- > > -- _ -- 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] Movistar sip Mexico
Have you followed the instructions in: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway ?? If possible try with a different ATA since it seems not all of them work fine with fax pass trough. Alyed 2013/11/21 Damian Gonzalez > Hi, > > I have Asterisk 10.12.1. I can not figure out the solution. > > Thank you for your help. > > Best Regards > > > On Thu, Nov 21, 2013 at 7:07 PM, Alyed wrote: > >> Which version of Asterisk are you using? >> >> According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless >> you are using Asterisk 10, there's quite some patching (or buying) you'll >> need to be doing. >> >> Alyed >> >> >> 2013/11/21 Bryant Zimmerman >> >>> Can you funnel them through a specific inbound dial context. Then force >>> a re-invite to g729? >>> >>> Thanks >>> >>> Bryant Zimmerman (ZK Tech Inc.) >>> 616-855-1030 Ext. 2003 >>> >>> >>> -- >>> *From*: "Damian Gonzalez" >>> *Sent*: Thursday, November 21, 2013 8:25 AM >>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < >>> asterisk-users@lists.digium.com> >>> *Subject*: Re: [asterisk-users] Movistar sip Mexico >>> >>> >>> Any posible solution? >>> >>> >>> On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner >>> wrote: >>> It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this). The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid. On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez >>> > wrote: > Hello, > > Thanks for the quickly response. I have only G729 in the peer but I > have t38pt_udptl= yes > > If I put t38pt_udptl=no , asterisk reject the call with 488 code. > > The problem is that Movistar send T38 codec in all calls and I need > ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have > only T38 I have to negociate a fax call. > > Thanks. > > > On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: > >> Think you only need to make sure you have in your sip.conf file these >> configs: >> >> [your-device-name] >> . >> . >> disallow=all >> allow=g729 >> . >> . >> >> >> Alyed >> >> 2013/11/20 Damian Gonzalez >> >>> Hello, >>> >>> I have a problem with movistar in Mexico with a sip calls. Movistar >>> send to me T38 and G729 in the INVITE and they say that I have to ignore >>> T38 and use G729 in the voice call. >>> >>> When a fax call is made Movistar send only T38 in the INVITE. >>> >>> Invite example: >>> >>> v=0 >>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 >>> s=sip call >>> c=IN IP4 192.168.1.2 >>> t=0 0 >>> m=audio 6370 RTP/AVP 18 101 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=ptime:20 >>> m=image 6372 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38FaxMaxBuffer:1100 >>> a=T38FaxMaxDatagram:612 >>> a=T38MaxBitRate:14400 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> How can I ignore T38 and use only G729 for this call?. >>> >>> Thanks for your help. >>> >>> Damian >>> >>> >>> -- >>> >>> >>> -- >>> _ >>> -- 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 > -- Kristian Kielhofner -- ___
Re: [asterisk-users] Call files without permission for asterisk to read
You could save the call file initially to "/var/spool/asterisk/tmp", then adjust the permissions as needed and necessary. Finally copy the call file into the "outgoing" directory. This also minimizes the chance that Asterisk tries to execute a partial file, although I don't know whether one still has to take care of issues like that. jg -- _ -- 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] Movistar sip Mexico
Which version of Asterisk are you using? According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless you are using Asterisk 10, there's quite some patching (or buying) you'll need to be doing. Alyed 2013/11/21 Bryant Zimmerman > Can you funnel them through a specific inbound dial context. Then force a > re-invite to g729? > > Thanks > > Bryant Zimmerman (ZK Tech Inc.) > 616-855-1030 Ext. 2003 > > > -- > *From*: "Damian Gonzalez" > *Sent*: Thursday, November 21, 2013 8:25 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > *Subject*: Re: [asterisk-users] Movistar sip Mexico > > > Any posible solution? > > > On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote: > >> It is possible that Asterisk requires an rtpmap even for static payload >> types (I'm not sure about this). The INVITE from your provider omits >> rtpmap for payload type 18 (G729) which is perfectly valid. >> >> >> On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez >> wrote: >> >>> Hello, >>> >>> Thanks for the quickly response. I have only G729 in the peer but I have >>> t38pt_udptl= yes >>> >>> If I put t38pt_udptl=no , asterisk reject the call with 488 code. >>> >>> The problem is that Movistar send T38 codec in all calls and I need >>> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have >>> only T38 I have to negociate a fax call. >>> >>> Thanks. >>> >>> >>> On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: >>> Think you only need to make sure you have in your sip.conf file these configs: [your-device-name] . . disallow=all allow=g729 . . Alyed 2013/11/20 Damian Gonzalez > Hello, > > I have a problem with movistar in Mexico with a sip calls. Movistar > send to me T38 and G729 in the INVITE and they say that I have to ignore > T38 and use G729 in the voice call. > > When a fax call is made Movistar send only T38 in the INVITE. > > Invite example: > > v=0 > o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 > s=sip call > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 6370 RTP/AVP 18 101 > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > m=image 6372 udptl t38 > a=T38FaxVersion:0 > a=T38FaxMaxBuffer:1100 > a=T38FaxMaxDatagram:612 > a=T38MaxBitRate:14400 > a=T38FaxRateManagement:transferredTCF > a=T38FaxUdpEC:t38UDPRedundancy > > How can I ignore T38 and use only G729 for this call?. > > Thanks for your help. > > Damian > > > -- > > > -- > _ > -- 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 >>> >> >> >> >> -- >> Kristian Kielhofner >> >> -- >> _ >> -- 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] Movistar sip Mexico
Hi, I have Asterisk 10.12.1. I can not figure out the solution. Thank you for your help. Best Regards On Thu, Nov 21, 2013 at 7:07 PM, Alyed wrote: > Which version of Asterisk are you using? > > According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless > you are using Asterisk 10, there's quite some patching (or buying) you'll > need to be doing. > > Alyed > > > 2013/11/21 Bryant Zimmerman > >> Can you funnel them through a specific inbound dial context. Then force a >> re-invite to g729? >> >> Thanks >> >> Bryant Zimmerman (ZK Tech Inc.) >> 616-855-1030 Ext. 2003 >> >> >> -- >> *From*: "Damian Gonzalez" >> *Sent*: Thursday, November 21, 2013 8:25 AM >> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < >> asterisk-users@lists.digium.com> >> *Subject*: Re: [asterisk-users] Movistar sip Mexico >> >> >> Any posible solution? >> >> >> On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner >> wrote: >> >>> It is possible that Asterisk requires an rtpmap even for static payload >>> types (I'm not sure about this). The INVITE from your provider omits >>> rtpmap for payload type 18 (G729) which is perfectly valid. >>> >>> >>> On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez >>> wrote: >>> Hello, Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes If I put t38pt_udptl=no , asterisk reject the call with 488 code. The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call. Thanks. On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: > Think you only need to make sure you have in your sip.conf file these > configs: > > [your-device-name] > . > . > disallow=all > allow=g729 > . > . > > > Alyed > > 2013/11/20 Damian Gonzalez > >> Hello, >> >> I have a problem with movistar in Mexico with a sip calls. Movistar >> send to me T38 and G729 in the INVITE and they say that I have to ignore >> T38 and use G729 in the voice call. >> >> When a fax call is made Movistar send only T38 in the INVITE. >> >> Invite example: >> >> v=0 >> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 >> s=sip call >> c=IN IP4 192.168.1.2 >> t=0 0 >> m=audio 6370 RTP/AVP 18 101 >> a=fmtp:18 annexb=yes >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> m=image 6372 udptl t38 >> a=T38FaxVersion:0 >> a=T38FaxMaxBuffer:1100 >> a=T38FaxMaxDatagram:612 >> a=T38MaxBitRate:14400 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxUdpEC:t38UDPRedundancy >> >> How can I ignore T38 and use only G729 for this call?. >> >> Thanks for your help. >> >> Damian >> >> >> -- >> >> >> -- >> _ >> -- 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 >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> -- >>> _ >>> -- 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 C
[asterisk-users] Dialing directly with username and password
It seems I am not finding the right syntax to dial directly using an
username/password. If I insert in my dialplan something like:
12345 => {
Dial(SIP/823*:5***@78.11.22.33/01342244560);
hangup();
}
Then I get:
[Nov 21 20:09:01] NOTICE[9069][C-0001689e]: chan_sip.c:29713
sip_request_call: Conflicting extension values given. Using
'823' and not '01342244560'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/823*:5@78.11.22.33/01342244560
[Nov 21 20:09:01] NOTICE[12287][C-0001689e]: chan_sip.c:22914
handle_response_invite: Failed to authenticate on INVITE to '"Leandro
Dardini" ;tag=as1c0d8470'
-- SIP/78.11.22.33-000144c3 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Which is the correct syntax to use to dial directly with username and
password?
Leandro
--
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Re: [asterisk-users] userfield not logged to CDR
>> I'm logging cdr via odbc to mysql. It seems that there is an intermittent >> problem where the CDR(userfield) isn't written to the database. Had this problem as well, needed to edit the cdr_odbc.conf and change loguniqueid to yes: loguniqueid=yes ; Required to use the userfield Doug -- _ -- 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] Monitor extension status
Hello, How do I track the status of an extension for socket? I'm trying to use the ExtensionState, but it is returning empty. thank you, Eduardo -- _ -- 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] Call files without permission for asterisk to read
On Thu, 21 Nov 2013, jg wrote: Finally copy the call file into the "outgoing" directory. This also minimizes the chance that Asterisk tries to execute a partial file... 'mv' not 'cp' Also, create the file on the same filesystem as the spool directory so 'mv' isn't silently 'promoted' to 'cp.' -- 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
Re: [asterisk-users] Call files without permission for asterisk to read
On Thu, 21 Nov 2013, Rizwan Hisham wrote: Hi all,I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? When you activate the secondary, 'touch' the files in the spool directory. -- 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
Re: [asterisk-users] Question about Management Interface
CDR wrote: I am trying to identify the module (*.so) that contains the Asterisk Management Interface, so as to set noload=XXX.so in modules.conf. Any idea? There is no module, it's provided as core functionality. Disabling it can be done in manager.conf -- Joshua Colp Digium, Inc. | Senior Software Developer 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] Call files without permission for asterisk to read
Looking at Eric Wieling's response and the wiki entry he mentioned, the precaution is still necessary. jg -- _ -- 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] Call files without permission for asterisk to read
Have you tried to restart asterisk after setting the correct permissions? HTH, Ioan On Thu, Nov 21, 2013 at 6:04 PM, Rizwan Hisham wrote: > Hi all, > I am syncing call files on my secondary asterisk server but without > permission to read for asterisk. So they should be executed when I grant > the right permissions (thats when my primary asterisk server crashes or > shutsdown somehow). But asterisk only tries to read the file at the time of > placing the file. So when i grant right permissions nothing happens. Is > there any workaround to this problem? > > I need to continue the execution of call files on secondary server if > primary server fails. The call files are suppose to retry for 45 mins if > the call does not get connected. > > Thanks in advance. > > -- > Best Ragards > Rizwan H Qureshi > > V: +971 (0) 528272154 > linkedin.com/in/rhqureshi > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files without permission for asterisk to read
Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Movistar sip Mexico
Any posible solution? On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote: > It is possible that Asterisk requires an rtpmap even for static payload > types (I'm not sure about this). The INVITE from your provider omits > rtpmap for payload type 18 (G729) which is perfectly valid. > > > On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez wrote: > >> Hello, >> >> Thanks for the quickly response. I have only G729 in the peer but I have >> t38pt_udptl= yes >> >> If I put t38pt_udptl=no , asterisk reject the call with 488 code. >> >> The problem is that Movistar send T38 codec in all calls and I need >> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have >> only T38 I have to negociate a fax call. >> >> Thanks. >> >> >> On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: >> >>> Think you only need to make sure you have in your sip.conf file these >>> configs: >>> >>> [your-device-name] >>> . >>> . >>> disallow=all >>> allow=g729 >>> . >>> . >>> >>> >>> Alyed >>> >>> 2013/11/20 Damian Gonzalez >>> Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian -- -- _ -- 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 >> > > > > -- > Kristian Kielhofner > > -- > _ > -- 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] Call files without permission for asterisk to read
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files Especially the parts about creating the files in a different directory and the parts about the scheduling call files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham Sent: Thursday, November 21, 2013 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call files without permission for asterisk to read Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- 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] Question about Management Interface
I am trying to identify the module (*.so) that contains the Asterisk Management Interface, so as to set noload=XXX.so in modules.conf. Any idea? -- _ -- 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] Movistar sip Mexico
Can you funnel them through a specific inbound dial context. Then force a re-invite to g729? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Damian Gonzalez" Sent: Thursday, November 21, 2013 8:25 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Movistar sip Mexico Any posible solution? On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote: It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this). The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid. On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez wrote: Hello, Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes If I put t38pt_udptl=no , asterisk reject the call with 488 code. The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call. Thanks. On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: Think you only need to make sure you have in your sip.conf file these configs: [your-device-name] . . disallow=all allow=g729 . . Alyed 2013/11/20 Damian Gonzalez Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian -- -- _ -- 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 -- Kristian Kielhofner -- _ -- 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
