Re: [asterisk-users] FAX with SIP
On 22/07/2011 5:43 AM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com mailto:davies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not well I tried a linksys spa 8000 and 2102 thru asterisk 1.8.3 1.8.4 1.6.2.16-19 sonus switch at itsp (012 israel) and no luck Cisco released updated firmware earlier this year for the SPA8000 SPA2102 which addresses T.38 problems. I have an SPA8800 which I was able to use T.38 mode to send faxes successfully, I recently updated Asterisk box and also updated to 1.8.5, haven't tested the SPA8800 with this config but I am expecting it will still work. The key to my success was to ensure the SPA8800 did not do a re-invite to the ISP for the RTP stream. 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] FAX with SIP
On Mon, Jul 18, 2011 at 07:58, Steve Davies davies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! -- _ -- 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] FAX with SIP
On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not. -- 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] FAX with SIP
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not well I tried a linksys spa 8000 and 2102 thru asterisk 1.8.3 1.8.4 1.6.2.16-19 sonus switch at itsp (012 israel) and no luck -- _ -- 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] FAX with SIP
On 07/21/2011 04:43 PM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com mailto:davies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not well I tried a linksys spa 8000 and 2102 thru asterisk 1.8.3 1.8.4 1.6.2.16-19 sonus switch at itsp (012 israel) and no luck We'd be happy to investigate why it failed, if you can capture the packet streams on both sides of Asterisk. Frequently, it's a configuration issue in at least one of the devices in the system. -- 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] FAX with SIP
On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming kpflem...@digium.com wrote: We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not. Do you care to share your exact configuration? I would love to reproduce it and demonstrate that T.38 in Asterisk indeed works. All I've ever managed to do is have it drop calls when the T.38 switchover is attempted. Does Asterisk even support T.38 AND NAT? -- _ -- 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] FAX with SIP
On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 04:43 PM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com mailto:davies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Correct. However it would be helpful to note T.38 support in Asterisk is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and try to send a fax. It won't work! We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not well I tried a linksys spa 8000 and 2102 thru asterisk 1.8.3 1.8.4 1.6.2.16-19 sonus switch at itsp (012 israel) and no luck We'd be happy to investigate why it failed, if you can capture the packet streams on both sides of Asterisk. Frequently, it's a configuration issue in at least one of the devices in the system. NP I'll get that for you I have spent days trying to get it to work with no luck -- 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-**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] FAX with SIP
On 11-07-21 05:52 PM, Joaquin Sosa wrote: On Thu, Jul 21, 2011 at 17:39, Kevin P. Flemingkpflem...@digium.com wrote: We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your It won't work statement with anything in particular. Given the variations of T.38 implementations that exist in ATAs, carrier networks and other places, *any* T.38 connection that involves implementations from more than one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved or not. Do you care to share your exact configuration? I would love to reproduce it and demonstrate that T.38 in Asterisk indeed works. All I've ever managed to do is have it drop calls when the T.38 switchover is attempted. Does Asterisk even support T.38 AND NAT? mnicholson recently added some tests[1] into the testsuite. [1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/fax/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
[asterisk-users] FAX with SIP
Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.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] FAX with SIP
On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Regards, 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] FAX with SIP
So Steve I looked this, but, i didn't understood the difference between enable T.38 and T.38 Gateway, this site ttp://www.voip-info.org/wiki/view/T.38 talk --Asterisk *1.6* support G.711 and T.38 FAX origination and termination. T.38 gateway features are still in development. -- I know that Asterisk 1.10-beta1 already work with T.38 gateway, but the ask is, i need T.38 gateway to fax works? and how i know if T.38 is enable? I put on sip.conf and sip_general_custom.conf the following entry... t38pt_udptl=yes Is right? Thank you!! 2011/7/18 Steve Davies davies...@gmail.com On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Regards, 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 -- Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.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] FAX with SIP
Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.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 -- _ -- 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] FAX with SIP
I resoundingly second that. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 18, 2011, at 11:12 PM, C F shma...@gmail.com wrote: Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.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 -- _ -- 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] Fax and SIP Device
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending fax out working well. In the mailing lists, i notice
some are using HT286 and it work.
Could someone share theirs experience and give some help?
E1
PSTN -- Asterisk (TE100P) HT486 Fax machine
zapata.conf
[channel]
context=Local
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
callwaiting=yes
usecallingpres=yes;
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=400
rxgain=0.0
txgain=0.0
immediate=no
faxdetect=both
musiconhold=default
extension.conf
[Incoming]
exten = 8003,1,Answer
exten = 8003,2,Wait(5)
exten = 8003,3,Macro(dial-sip,1300,${LONGTIMEOUT})
exten = fax,1,Dial(SIP/1300,30)
exten = fax,2,Congestion
exten = fax,102,Congestion
sip.conf
[1300]
type=friend
host=dynamic
username=xxx
secret=xxx
nat=yes
context=Local
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
On the other hand, i was tired with getting the same error when
compiling rxfax/txfax.
spandsp-0.0.2pre18 and spandsp-0.0.1
gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c
app_rxfax.c: In function `phase_e_handler':
app_rxfax.c:77: error: structure has no member named `callerid'
make[1]: *** [app_rxfax.o] Error 1
Best Regards
Khng
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Re: [Asterisk-Users] Fax and SIP Device
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending fax out working well. In the mailing lists, i notice
some are using HT286 and it work.
Could someone share theirs experience and give some help?
E1
PSTN -- Asterisk (TE100P) HT486 Fax machine
zapata.conf
[channel]
context=Local
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
callwaiting=yes
usecallingpres=yes;
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=400
rxgain=0.0
txgain=0.0
immediate=no
faxdetect=both
musiconhold=default
extension.conf
[Incoming]
exten = 8003,1,Answer
exten = 8003,2,Wait(5)
exten = 8003,3,Macro(dial-sip,1300,${LONGTIMEOUT})
exten = fax,1,Dial(SIP/1300,30)
exten = fax,2,Congestion
exten = fax,102,Congestion
sip.conf
[1300]
type=friend
host=dynamic
username=xxx
secret=xxx
nat=yes
context=Local
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
On the other hand, i was tired with getting the same error when
compiling rxfax/txfax.
spandsp-0.0.2pre18 and spandsp-0.0.1
gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c
app_rxfax.c: In function `phase_e_handler':
app_rxfax.c:77: error: structure has no member named `callerid'
make[1]: *** [app_rxfax.o] Error 1
In zapta.conf you have:
[channel]
context=Local
which sends incoming calls to the Local context.
However, your extensions.conf fax sample is in context [Incoming].
Either that is your issue, or, we'll need to see the [Local]
context.
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[Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following messages start to appear (about 100 of them) Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 127 received After that asterisk gets totaly confused: Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 1 received Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (15)? Nov 23 16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format: Unable to find a path from G723 to ALAW Nov 23 16:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 == Spawn extension (macro-enumcall, s, 211) exited non-zero on 'SIP/sip.westend.com-082fd1b8' in macro 'enumcall' == Spawn extension (xxx, 911879, 7) exited non-zero on 'SIP/sip.westend.com-082fd1b8' -- Executing NoOp(SIP/sip.westend.com-082fd1b8, ) in new stack cdr_odbc: Query Successful! Then the call gets hung up. I cannot explain why this happens. I would have explanations for the fax-machines not able to synchronize or faxes not being transmitted correctly, as the communication is SIP only, but this seems a bit strange to me. I can't even tell, if my asterisk produces these messages, or the other side (aka PSTN-Gateway). If anyone can bring some light into this behavior I would be very greatful. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. I'm interested in T.38 support too, so if anybody could explain why * can't just pass theese packets (as i undrstand there is no need foe recoding etc.) I would be very appreciative. Are anybody currently working on T.38 support for * ? I don't mean T.38 support on zap interfaces, just passing T.38 packets trouth asterisk -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following messages start to appear (about 100 of them) Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 127 received After that asterisk gets totaly confused: Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 1 received Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (15)? Nov 23 16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format: Unable to find a path from G723 to ALAW Nov 23 16:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 == Spawn extension (macro-enumcall, s, 211) exited non-zero on 'SIP/sip.westend.com-082fd1b8' in macro 'enumcall' == Spawn extension (xxx, 911879, 7) exited non-zero on 'SIP/sip.westend.com-082fd1b8' -- Executing NoOp(SIP/sip.westend.com-082fd1b8, ) in new stack cdr_odbc: Query Successful! Then the call gets hung up. I cannot explain why this happens. I would have explanations for the fax-machines not able to synchronize or faxes not being transmitted correctly, as the communication is SIP only, but this seems a bit strange to me. I can't even tell, if my asterisk produces these messages, or the other side (aka PSTN-Gateway). If anyone can bring some light into this behavior I would be very greatful. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Ltticher Strae 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Hello! Elman Efendiyev wrote: Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. No, I'm not trying to send the fax over T.38, I am trying to send it in the voice path by using the G711 alaw codec. This should work, I think, but it doesn't. Best regards Kai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Log you posted looks like sitll T.38 problemm Which gates you use? Gateways able to support T.38 will try to use it by default no matter what codec in use. I'd suggest check gateway setup if T.38 is completely disabled Fax call with G711 passtrouth (without T.38) havent any difference comparing to voice call, you can (and probaby will) have troubles with fax transmission (quality, line drops etc) but not with * complain about codecs. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello! Elman Efendiyev wrote: Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. No, I'm not trying to send the fax over T.38, I am trying to send it in the voice path by using the G711 alaw codec. This should work, I think, but it doesn't. Best regards Kai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
At 21:20 16-11-2003 -0600, you wrote: You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Be advised this item regards the ATA186-L series, and the issues should be resolved with the I1 and I2 models. (the L series are End Of Life and have been since mid-2002). Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Dave Weis wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
At 20:34 14-11-2003 -0600, you wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. I have gotten this to work, but it seems to be very dependant on your infrastructure. Fax over alaw or ulaw clear channels will work as long as the latency of the network is within acceptable range. Give it a try :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
Am Sam, 2003-11-15 um 11.22 schrieb Florian Overkamp: At 20:34 14-11-2003 -0600, you wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. I have gotten this to work, but it seems to be very dependant on your infrastructure. Fax over alaw or ulaw clear channels will work as long as the latency of the network is within acceptable range. Give it a try :-) Florian On a LAN this should work without problems. I am even faxing over the wild internet with ping times between 100 and 120 ms. Both locations have an AVM Fritz card connected to an internal S0 bus of an isdn pbx and run chan_capi. best regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over SIP alaw/ulaw
Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FAX over SIP
Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line - mp108fxo - Asterisk -- mp108fxs --- fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid extension ff in context ). 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call to this extension but when the fax answers the call is dropped ( don't have here the SIP debug output ) but seems that when * tries to make the bridge the mp108 fxo sends a BYE. 3) I dialed in to * with a phone ( external line and internal extension ) and dial extension for the fax and i cann hear the fax, so the call is not dropped, the bridge is established successfully. 4) If I pickup the call on the fax machine ( it has a phone set ) and then pressed the 'start' button to start de fax receiver, then, the two faxes talked to each other and the fax is received well. Seems that the problem is only when the fax answer automatically ( could be the tones the receiving fax plays ? ), the same problem happens when i try to use hylafax to receive the fax. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over SIP
On Thu, 2003-09-04 at 01:18, Ing. Angel Gomez Garcia wrote: Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line - mp108fxo - Asterisk -- mp108fxs --- fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid extension ff in context ). 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call to this extension but when the fax answers the call is dropped ( don't have here the SIP debug output ) but seems that when * tries to make the bridge the mp108 fxo sends a BYE. 3) I dialed in to * with a phone ( external line and internal extension ) and dial extension for the fax and i cann hear the fax, so the call is not dropped, the bridge is established successfully. 4) If I pickup the call on the fax machine ( it has a phone set ) and then pressed the 'start' button to start de fax receiver, then, the two faxes talked to each other and the fax is received well. By listening to #4 it sounds like you answered your own question, yes it is possible. Now you need to find out why in #2 that it sends a bye. My guess is that you have a difference in the dial command options that keep asterisk listening to the line when dialing the extension that isn't there on the ff extension. This may have asterisk trying to issue a reinvite to connect the call legs together without asterisk in the middle. This is causing the BYE, and then everything fall apart. Maybe you need to make sure the canreinvite is turned off for this device in the sip.conf and try some more. Seems that the problem is only when the fax answer automatically ( could be the tones the receiving fax plays ? ), the same problem happens when i try to use hylafax to receive the fax. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
At 15:30 26-6-2003 -0300, you wrote: I've tested ATA186 with a cisco827 as the H323 (or SIP) gateway and I could transmite the fax without problem. I get erros when sending faxes only when I user asterisk. :~ any tips? I imagine the Cisco stuff uses T30/T38 amongst eachother. Asterisk (I think) doesn't support T30/T38, so it is much more prone to errors if the setup is not just perfect. Maybe someone has nifty tools to debug faxing ? Out of my league for sure.. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
On Thu, 26 Jun 2003 09:01:21 +0200 Florian Overkamp [EMAIL PROTECTED] wrote: Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. I've tested with both on the localnet (same ethernet hub) and I still get errors on the fax machine. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
I've never tried it with SIP, but I have faxed between to asterisk boxes on the same network via IAX and IAX2. The secret was to set the codec to ulaw or alaw. (Certain codecs, such as GSM, compress the data too much for the fax machines to be able to communicate effectively.) Jared On Thu, 2003-06-26 at 08:10, Eduardo Goncalves wrote: On Thu, 26 Jun 2003 09:01:21 +0200 Florian Overkamp [EMAIL PROTECTED] wrote: Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. I've tested with both on the localnet (same ethernet hub) and I still get errors on the fax machine. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
Have you tried limiting your fax machines to a lower baud rate like 9600. I know on Vonage this seems to help. - Original Message - From: Eduardo Goncalves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 26, 2003 10:10 AM Subject: Re: [Asterisk-Users] Fax and SIP On Thu, 26 Jun 2003 09:01:21 +0200 Florian Overkamp [EMAIL PROTECTED] wrote: Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. I've tested with both on the localnet (same ethernet hub) and I still get errors on the fax machine. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
REMOVE - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 26, 2003 4:01 AM Subject: Re: [Asterisk-Users] Fax and SIP At 16:01 25-6-2003 -0300, you wrote: Hi list, I have the following scenario, and want to know what I have to do to transmit faxes trought this link: |cisco-ata186|sip-|asterisk|---EM alaw link-PSTN The codec used is g711a. When I try to transmit a fax I receive a TX FUNCTION WAS NOT COMLETED on the fax machine connected to cisco ATA186. Could someone help me? Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
Jim == Jim Flagg [EMAIL PROTECTED] writes: Jim Have you tried limiting your fax machines to a lower baud rate Jim like 9600. I know on Vonage this seems to help. Speaking of which, IIRC the docs for the ata mention that fax at greater than 9600 is b0rked up to a recent firmware release. You (the OP) may need to upgrade the ata to get it to work. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
On 26 Jun 2003 12:53:40 -0400 James H. Cloos Jr. [EMAIL PROTECTED] wrote: Jim == Jim Flagg [EMAIL PROTECTED] writes: Jim Have you tried limiting your fax machines to a lower baud rate Jim like 9600. I know on Vonage this seems to help. Speaking of which, IIRC the docs for the ata mention that fax at greater than 9600 is b0rked up to a recent firmware release. You (the OP) may need to upgrade the ata to get it to work. I've tested ATA186 with a cisco827 as the H323 (or SIP) gateway and I could transmite the fax without problem. I get erros when sending faxes only when I user asterisk. :~ any tips? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax and SIP
Hi list, I have the following scenario, and want to know what I have to do to transmit faxes trought this link: |cisco-ata186|sip-|asterisk|---EM alaw link-PSTN The codec used is g711a. When I try to transmit a fax I receive a TX FUNCTION WAS NOT COMLETED on the fax machine connected to cisco ATA186. Could someone help me? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
