Re: [Asterisk-Users] iax / realtime problems
Hi Mat, Did the following: 1. Upgraded to new CVS HEAD version CVS-NHEAD-04/11/05-16:08:03 On the Makefile, enabled the ff: # Optional debugging parameters DEBUG_THREADS = -DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS MALLOC_DEBUG = -include $(PWD)/include/asterisk/astmm.h I cannot seem to enable pg on this line in Makefile #Include debug symbols in the executables (-g) and profiling info (-pg) DEBUG=-g #-pg I get error below when I do make valgrind gcc: -pg and -fomit-frame-pointer are incompatible I skip enabling pg and continue with make clean and make valgrind. gdb backtrace still gives vague output: (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0048087a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00304420 in ?? () #5 0x0100 in ?? () #6 0x0049f900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00304580 in ?? () #8 0x00599605 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x0049f8fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00304840 in ?? () #11 0x0048074c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so Still not clear, any pointers to make the backtrace more verbose? On Mon, 2005-04-11 at 00:05, Matthew Boehm wrote: In order for this to be helpful, you need to recompile with make valgrind and edit your Makefile and turn on all the debugging stuff. -Matthew From: Paul P. Pongco [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 9 Apr 2005 15:13:55 +0800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] iax / realtime problems Hi Mat, I can easily replicate the problem. I just put an entry on the iax table for mysql, fire up iax soft client and BOOM .. asterisk core dumps. What's weird is sip is working fine using realtime. Here is a gdb backtrace. Not really a programmer. Hope someone helps. Thanks. #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0031187a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00b19340 in ?? () #5 0x0100 in ?? () #6 0x00330900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00b19480 in ?? () #8 0x0082d5d6 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x003308fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00b19720 in ?? () #11 0x0031174c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #12 0x in ?? () On Apr 8, 2005 8:53 PM, Matt Schulte [EMAIL PROTECTED] wrote: I've never actually core dumped but I *have* been able to hang asterisk a couple times, I believed my problem was when I lost my mysql connection. Why it lost connection is a mystery, the servers are on the same testswitch. :/ I forgot which head ver it was, a couple weeks ago. -Original Message- From: Paul P. Pongco [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax / realtime problems Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes
Re: [Asterisk-Users] iax / realtime problems
Hi Mat, I can easily replicate the problem. I just put an entry on the iax table for mysql, fire up iax soft client and BOOM .. asterisk core dumps. What's weird is sip is working fine using realtime. Here is a gdb backtrace. Not really a programmer. Hope someone helps. Thanks. #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0031187a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00b19340 in ?? () #5 0x0100 in ?? () #6 0x00330900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00b19480 in ?? () #8 0x0082d5d6 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x003308fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00b19720 in ?? () #11 0x0031174c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #12 0x in ?? () On Apr 8, 2005 8:53 PM, Matt Schulte [EMAIL PROTECTED] wrote: I've never actually core dumped but I *have* been able to hang asterisk a couple times, I believed my problem was when I lost my mysql connection. Why it lost connection is a mystery, the servers are on the same testswitch. :/ I forgot which head ver it was, a couple weeks ago. -Original Message- From: Paul P. Pongco [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax / realtime problems Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes What could be causing this? Anyone seen this problem before? Help would be appreciated. Thanks. -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax / realtime problems
Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes What could be causing this? Anyone seen this problem before? Help would be appreciated. Thanks. -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc sip
Hello, Can anyone point me to any documentation with regards to using sip_friends on astcc. astcc already working on our test * server but im trying to figure out how to sql-ize sip user config. I have thought of using Asterisk Realtime but is not yet available on stable release. Appreciate any pointers on this subject. Thanks! -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: asterisk+radius
Hello, Im actually deciding if I will use asterisk+radius for AAA purposes or use logging directly to mysql and using Asterisk+RealTime to store SIP users to mysql also. Question is, what's the best way to disconnect a user, if for example, he runs out of credits. thanks. On Fri, 2005-03-18 at 02:33, izo wrote: set asterisk to log into database directly via there are mysql , postgresql and odbc drivers available. You dont need radius at all, for billing and accounting all u need is a frontend to database On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote: Oh this is sad.. I'm familiar with radius.. and was hoping to be able to use asterisk with freeradius to be able to do call accounting and billing.. so you're telling me this is now not a good idea? Am I better off (for now) parsing the csv report each month? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: asterisk+radius
oops never mind, ill just read on astcc On Fri, 2005-03-18 at 14:10, Paul P. Pongco wrote: Hello, Im actually deciding if I will use asterisk+radius for AAA purposes or use logging directly to mysql and using Asterisk+RealTime to store SIP users to mysql also. Question is, what's the best way to disconnect a user, if for example, he runs out of credits. thanks. On Fri, 2005-03-18 at 02:33, izo wrote: set asterisk to log into database directly via there are mysql , postgresql and odbc drivers available. You dont need radius at all, for billing and accounting all u need is a frontend to database On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote: Oh this is sad.. I'm familiar with radius.. and was hoping to be able to use asterisk with freeradius to be able to do call accounting and billing.. so you're telling me this is now not a good idea? Am I better off (for now) parsing the csv report each month? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weird outbound problem through broadvoice (new)
Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a snapshot of my sip.conf register = [EMAIL PROTECTED]:PP:[EMAIL PROTECTED] [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromuser=UU fromdomain=sip.broadvoice.com secret=PP username=UU port=5060 dtmfmode=inband dtmf=inband insecure=very context=incoming authname=UU canreinvite=no qualify=no nat=no extensions.conf [outgoing] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX, 2, congestion() exten = _1NXXNXX, 102, busy() A portion of sip debug during successful calls (calling broadvoice support) Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines CLI Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED];tag=SD58a8499-104694000-1110784950009 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported: 100rel,timer Contact: sip:[EMAIL PROTECTED]:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp Remote-Party-ID: Auto Attendant PrimaryAttendantsip:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e;screen=yes;party=called;privacy=off;id-type=subscriber Content-Length: 0 A portion of sip debug during unsuccessful calls, where T is the target phone number Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username=UU, realm=BroadWorks, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=1110785211206, response=f68a31735aec843b9ef68b7909fcf178, opaque= Content-Length: 0 (no NAT) to 147.135.8.128:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Transmitting (no NAT): SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c From: sip:[EMAIL PROTECTED];tag=9d9e03fd7b4508e9 To: sip:[EMAIL PROTECTED];tag=as79fd7936 Call-ID: [EMAIL PROTECTED] CSeq: 7327 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to x.x.x.x:5060 Asterisk box not behind firewall. No iptables filters either. It seems that asterisk is sending CANCEL due to call timeout after the 2nd 100 Trying during INVITE message flow. I am not sure what is causing the timeout. Anyone experienced this before? Tried using ethereal to debug the problem deeply, but I can only see the same flow as the sip debug. Hoping for your assistance. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] weird outbound problem through broadvoice (new)
Hello, I changed my asterisk to the recently posted software on CVS (Asterisk CVS-v1-0-03/15/05-12:11:02). Problem still persists. What is weird here is I can dial certain numbers (broadvoice support number works) but cant on others. Checked the SIP call flow via ethereal and I can see Im sending and receiving invites from the same broadvoice server (147.135.8.128) w/c is what I have mapped sip.broadvoice.com to at /etc/hosts. Any other way I can debug this? Thanks. On Mon, 2005-03-14 at 17:40, Paul P. Pongco wrote: Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a snapshot of my sip.conf register = [EMAIL PROTECTED]:PP:[EMAIL PROTECTED] [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromuser=UU fromdomain=sip.broadvoice.com secret=PP username=UU port=5060 dtmfmode=inband dtmf=inband insecure=very context=incoming authname=UU canreinvite=no qualify=no nat=no extensions.conf [outgoing] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX, 2, congestion() exten = _1NXXNXX, 102, busy() A portion of sip debug during successful calls (calling broadvoice support) Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines CLI Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED];tag=SD58a8499-104694000-1110784950009 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported: 100rel,timer Contact: sip:[EMAIL PROTECTED]:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp Remote-Party-ID: Auto Attendant PrimaryAttendantsip:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e;screen=yes;party=called;privacy=off;id-type=subscriber Content-Length: 0 A portion of sip debug during unsuccessful calls, where T is the target phone number Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username=UU, realm=BroadWorks, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=1110785211206, response=f68a31735aec843b9ef68b7909fcf178, opaque= Content-Length: 0 (no NAT) to 147.135.8.128:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Transmitting (no NAT): SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c From: sip:[EMAIL PROTECTED];tag=9d9e03fd7b4508e9 To: sip:[EMAIL PROTECTED];tag=as79fd7936 Call-ID: [EMAIL PROTECTED] CSeq: 7327 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to x.x.x.x:5060 Asterisk box not behind firewall. No iptables filters either. It seems that asterisk is sending CANCEL due to call timeout after the 2nd 100 Trying during INVITE message flow. I am not sure what is causing the timeout. Anyone experienced this before? Tried using ethereal to debug the problem deeply, but I can only see the same flow as the sip debug. Hoping for your assistance. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple sip phones behind firewall
Hello List, Can you please point me to the right resources on making multiple sip phones behind a firewall w/ private address work with asterisk w/c is on a public network. I have seen STUN on the grandstream and Xtunnels on X-lite. What is most deployed by members here with similar setups? Thanks. -- Cheers, Paul P. Pongco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
