Problems with sip provider
I am using 3 different SIP providers. 2 of them work just fine, but I have never managed to get the third one, telfort (NL) to work with Freeswitch. Here is my gateway definition: <gateway name="telfort"> <param name="username" value="020NNNNNNN"/> <param name="auth-username" value="020NNNNNNN"/> <param name="from-user" value="020NNNNNNN"/> <param name="realm" value="tel.telefoniedienst.nl"/> <param name="from-domain" value="tel.telefoniedienst.nl"/> <param name="proxy" value="sip.telefoniedienst.nl"/> <param name="password" value="XXXXXXX"/> </gateway> sofia status shows: external::telfort gateway sip:020NNNNNNN@sip.telefoniedienst.nl REGED but when I try to dial out on that provider, I get: 2013-02-17 18:04:19.216205 [NOTICE] sofia.c:6385 Hangup sofia/external/06MMMMMMMM [CS_EXCHANGE_MEDIA] [CALL_REJECTED] The strange thing is that everything works OK when I configure the parameters into the ADSL modem/gateway - a ZyXEL P-2601HN-F1: SIP Server Address: sip.telefoniedienst.nl Port: 5060 REGISTER Server Address: sip.telefoniedienst.nl Port: 5060 SIP Service Domain: tel.telefoniedienst.nl Port: 5060 RTP Port Range: 16384-32767 Enable Early Media Account Number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXX Compression Type: G.711ALaw, G.729, G.722 Active G.168 Echo Cancellation so somehow I am not mapping that information properly into the Freeswitch conf... Julf
If they are rejecting it, I'd ask them why. Are you bridging to sofia/gateway/telfort ? Also, if you changed the password on the gateway, did you killgw the old load and restart it? reloadxml doesn't restart gateways that are already connected. -Avi On Sun, Feb 17, 2013 at 7:13 PM, Johan Helsingius <julf@julf.com> wrote:
I am using 3 different SIP providers. 2 of them work just fine, but I have never managed to get the third one, telfort (NL) to work with Freeswitch.
Here is my gateway definition:
<gateway name="telfort"> <param name="username" value="020NNNNNNN"/> <param name="auth-username" value="020NNNNNNN"/> <param name="from-user" value="020NNNNNNN"/> <param name="realm" value="tel.telefoniedienst.nl"/> <param name="from-domain" value="tel.telefoniedienst.nl"/> <param name="proxy" value="sip.telefoniedienst.nl"/> <param name="password" value="XXXXXXX"/> </gateway>
sofia status shows:
external::telfort gateway sip:020NNNNNNN@sip.telefoniedienst.nl REGED
but when I try to dial out on that provider, I get:
2013-02-17 18:04:19.216205 [NOTICE] sofia.c:6385 Hangup sofia/external/06MMMMMMMM [CS_EXCHANGE_MEDIA] [CALL_REJECTED]
The strange thing is that everything works OK when I configure the parameters into the ADSL modem/gateway - a ZyXEL P-2601HN-F1:
SIP Server Address: sip.telefoniedienst.nl Port: 5060 REGISTER Server Address: sip.telefoniedienst.nl Port: 5060 SIP Service Domain: tel.telefoniedienst.nl Port: 5060 RTP Port Range: 16384-32767 Enable Early Media Account Number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXX Compression Type: G.711ALaw, G.729, G.722 Active G.168 Echo Cancellation
so somehow I am not mapping that information properly into the Freeswitch conf...
Julf
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If they are rejecting it, I'd ask them why.
I did. "What equipment are you using?" "freeswitch" "we don't support that"
Are you bridging to sofia/gateway/telfort ?
Yes.
Also, if you changed the password on the gateway, did you killgw the old load and restart it? reloadxml doesn't restart gateways that are already connected.
I restarted freeswitch. Julf
Can you provide SIP trace of the call on FS and ZyXEL? That may provide some details of the problem. I remembered a similar problem i faced with an ITSP in Pakistan, and after tracing problem for a couple days i found that they were using User Agent filter to allow calls only from their "preferred" devices. So, we had to hack that to get it working with FS, http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#user-agent-string Thank you. On Sun, Feb 17, 2013 at 5:33 PM, Johan Helsingius <julf@julf.com> wrote:
If they are rejecting it, I'd ask them why.
I did.
"What equipment are you using?" "freeswitch" "we don't support that"
Are you bridging to sofia/gateway/telfort ?
Yes.
Also, if you changed the password on the gateway, did you killgw the old load and restart it? reloadxml doesn't restart gateways that are already connected.
I restarted freeswitch.
Julf
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Can you provide SIP trace of the call on FS and ZyXEL? That may provide some details of the problem.
On FS, no prob, a bit harder on the ZyXEL, as I don't have access to the outgoing (ADSL) side of it.
I remembered a similar problem i faced with an ITSP in Pakistan, and after tracing problem for a couple days i found that they were using User Agent filter to allow calls only from their "preferred" devices. So, we had to hack that to get it working with FS,
Ouch! Nasty! But in the case of telfort.nl, they do support 3rd party hardware. This is what their guide says for the settings: SIP proxy: sip.telefoniedienst.nl SIP service location: sip.telefoniedienst.nl SIP signaling port: UDP 5060 SIP register service location: tel.telefoniedienst.nl SIP register signaling port: UDP 5060 SIP domain: tel.telefoniedienst.nl URL type coding: SIP Registration interval: 6000 seconds Session expires: 1800 seconds Reserved RTP port range: UDP 16384-32767 Codecs: First negotiated G.729 secondly negotiated G.711 a-law type DTMF mode: PCM Sending INVITE Anonymous: Not permitted Echo Cancellation G.168: In use Voice Activation Detection: Not in use SIP SUBSCRIBES messages: Not permitted Early Media acknowledge: Not in Use Julf
"What equipment are you using?" "freeswitch" "we don't support that"
A generic SIP standards-compliant client... They may be trying to answer the wrong question. You don't need them to tell you how to configure your device to talk to them (a FS specific question), you want them to tell you why the authentication failed (specific to their end). Once we know that it should be possible to know what needs to be adjusted. -Steve On 17 February 2013 17:33, Johan Helsingius <julf@julf.com> wrote:
If they are rejecting it, I'd ask them why.
I did.
"What equipment are you using?" "freeswitch" "we don't support that"
Are you bridging to sofia/gateway/telfort ?
Yes.
Also, if you changed the password on the gateway, did you killgw the old load and restart it? reloadxml doesn't restart gateways that are already connected.
I restarted freeswitch.
Julf
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You don't need them to tell you how to configure your device to talk to them (a FS specific question), you want them to tell you why the authentication failed (specific to their end). Once we know that it should be possible to know what needs to be adjusted.
Indeed. But the problem is that I can't get past their (useless) first level helpdesk to talk to the people who might actually have access to their logs. Julf
Ask them for a list of devices that they support. Once you have that we can probably help you figure out how to fool them into thinking that they are talking to one of those "supported" devices. In cases where the provider is so totally foolish like this you have two choices: * Use their foolishness against them and make them think your FreeSWITCH server is really something else * Switch providers I know that's probably not what you want to hear but it is indeed the truth. -MC On Sun, Feb 17, 2013 at 10:48 PM, Johan Helsingius <julf@julf.com> wrote:
You don't need them to tell you how to configure your device to talk to them (a FS specific question), you want them to tell you why the authentication failed (specific to their end). Once we know that it should be possible to know what needs to be adjusted.
Indeed. But the problem is that I can't get past their (useless) first level helpdesk to talk to the people who might actually have access to their logs.
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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+1. We suffered a similar problem with our first round of providers, it took us nearly 4 months to get them to fix it, and even after they fixed it, we found other problems. It's totally your choice if you want to continue trying to get it fixed, but most of the time, it's not worth the time/effort. My two cents at least. Cal On Mon, Feb 18, 2013 at 6:34 PM, Michael Collins <msc@freeswitch.org> wrote:
Ask them for a list of devices that they support. Once you have that we can probably help you figure out how to fool them into thinking that they are talking to one of those "supported" devices. In cases where the provider is so totally foolish like this you have two choices: * Use their foolishness against them and make them think your FreeSWITCH server is really something else * Switch providers
I know that's probably not what you want to hear but it is indeed the truth.
-MC
On Sun, Feb 17, 2013 at 10:48 PM, Johan Helsingius <julf@julf.com> wrote:
You don't need them to tell you how to configure your device to talk to them (a FS specific question), you want them to tell you why the authentication failed (specific to their end). Once we know that it should be possible to know what needs to be adjusted.
Indeed. But the problem is that I can't get past their (useless) first level helpdesk to talk to the people who might actually have access to their logs.
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Is it possible to get a SIP trace of the calls from the gateway and FS and compare them? -Steve On 19 February 2013 09:23, Johan Helsingius <julf@julf.com> wrote:
It's totally your choice if you want to continue trying to get it fixed, but most of the time, it's not worth the time/effort.
Have to agree - just annoying that it does work when using their ADSL gateway :(
Julf
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Well, at least provide a SIP trace which INVITE request FS sends to the provider and which response the provider sends back. On Tuesday 19 February 2013 10:56:42 Johan Helsingius wrote:
Is it possible to get a SIP trace of the calls from the gateway and FS and compare them?
Unfortunately there is no way to get into the traffic between the gateway and the ADSL connection, as it also contains the ADSL modem.
Julf
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Well, at least provide a SIP trace which INVITE request FS sends to the provider and which response the provider sends back.
Here: http://pastebin.freeswitch.org/20605 I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" <sip:0000000000@tel.telefoniedienst.nl> in the INVITE? Julf
To get a SIP trace from ADSL, is a little tricky but if you manage it here are the steps which may help, 1. Setup an OpenSIPS server as stateless proxy. 2. In ADSL set that server as Outbound Proxy. 3. Make a call, it will pass through OpenSIPS where you can do a SIP trace. 4. You can even modify SIP requests and replies there and get a hint of their restrictions. For example change user-agent in INVITE from ADSL to FS's user-agent, and see what happens? Does the INVITE gets rejected? etc. etc. Some security paranoid providers even do restriction based on your ADSL device's MAC address, which is issued and activated by them. Still there are ways to get into their network. All that matters is how badly you need that provider. ;-) Thank you. On Tue, Feb 19, 2013 at 2:03 PM, Johan Helsingius <julf@julf.com> wrote:
Well, at least provide a SIP trace which INVITE request FS sends to the provider and which response the provider sends back.
Here:
http://pastebin.freeswitch.org/20605
I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" <sip:0000000000@tel.telefoniedienst.nl> in the INVITE?
Julf
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-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
To get a SIP trace from ADSL, is a little tricky but if you manage it here are the steps which may help,
1. Setup an OpenSIPS server as stateless proxy.
Yes, that would probably do it, but if I really have to go trough all that trouble, I think I might just dump that provider.
Some security paranoid providers even do restriction based on your ADSL device's MAC address, which is issued and activated by them. Still there are ways to get into their network. All that matters is how badly you need that provider. ;-)
They do state that they do allow third-party equipment and systems, they just don't provide any customer support for them, so I assume they aren't blocking based on MAC address. But you never know... Julf
I suspect the problem is codec-related. The provider behaves badly when g729 is not offered. On Tuesday 19 February 2013 14:03:12 Johan Helsingius wrote:
Well, at least provide a SIP trace which INVITE request FS sends to the provider and which response the provider sends back.
Here:
http://pastebin.freeswitch.org/20605
I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" <sip:0000000000@tel.telefoniedienst.nl> in the INVITE?
Julf
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g729 license will never hurt. On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote:
I suspect the problem is codec-related. The provider behaves badly when g729 is not offered.
Ah. And I guess the way to deal with that is to purchase a g729 license?
Julf
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No, 603 is generic error, codec related problems are usually 488 Not Acceptable Here. Also they are using Digitalk SBC which has no codec based restriction as far as i know. http://www.digitalk.com/carrier/Session_Border_Controller/ Thank you. On Tue, Feb 19, 2013 at 3:05 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
g729 license will never hurt.
On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote:
I suspect the problem is codec-related. The provider behaves badly when g729 is not offered.
Ah. And I guess the way to deal with that is to purchase a g729 license?
Julf
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-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
In theory and according to the standards yes, but they could send anything they want in practice. It'd be incorrect and annoying, but not impossible. As for the SBC, the limitation could be in their configuration rather than the software, or further along the call path. -Steve On 19 February 2013 14:31, Muhammad Shahzad <shaheryarkh@gmail.com> wrote:
No, 603 is generic error, codec related problems are usually 488 Not Acceptable Here.
Also they are using Digitalk SBC which has no codec based restriction as far as i know.
http://www.digitalk.com/carrier/Session_Border_Controller/
Thank you.
On Tue, Feb 19, 2013 at 3:05 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org> wrote:
g729 license will never hurt.
On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote:
I suspect the problem is codec-related. The provider behaves badly when g729 is not offered.
Ah. And I guess the way to deal with that is to purchase a g729 license?
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Just talked to a friend, he says though Digitalk SBC is extremely configurable, but most providers specially in NL (due to its telecom and legal interception laws etc.), force customers to provide both caller and callee ids correctly. So, you need at least to make sure, 1. Destination number is in correct format, usually E.164 number but provider may choose differently, e.g. give you some dial prefix to dial out etc. 2. You are using Remote Part ID in correct format, usually the format is same as destination number format. 3. If using anonymous caller id then make sure to set Privacy, P-Asserted Identity and / or P-Preferred Identity. Try these if work then good otherwise just dump the carrier. Thank you. On Tue, Feb 19, 2013 at 3:58 PM, Johan Helsingius <julf@julf.com> wrote:
No, 603 is generic error, codec related problems are usually 488 Not Acceptable Here.
Right - and I verified it by installing a g729 license and the mod_com_g729 module - no difference.
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
Muhammad,
Just talked to a friend, he says though Digitalk SBC is extremely configurable, but most providers specially in NL (due to its telecom and legal interception laws etc.), force customers to provide both caller and callee ids correctly. So, you need at least to make sure,
1. Destination number is in correct format, usually E.164 number but provider may choose differently, e.g. give you some dial prefix to dial out etc. 2. You are using Remote Part ID in correct format, usually the format is same as destination number format.
Ah! Yes! Setting caller-id-in-from to true and effective-outbound-caller-id-number to the proper E.164 format of the assigned number did it! Many thanks! Julf
Ah! Yes!
Setting caller-id-in-from to true and effective-outbound-caller-id-number to the proper E.164 format of the assigned number did it!
In other words, carriers in the USA don't have a monopoly on their support
reps not knowing how their own systems work! -MC
Many thanks!
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
On Tue, Feb 19, 2013 at 10:40 AM, Johan Helsingius <julf@julf.com> wrote:
In other words, carriers in the USA don't have a monopoly on their support reps not knowing how their own systems work!
They are probably all sharing the same call center in some low-wage country. Made possible by VoIP, of course... :)
Haha - we're our own worst enemy - the OSS telephony geeks are subverted by the work of their own hands! -MC
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Great. Glad to know it worked out for you. Thank you. On Tue, Feb 19, 2013 at 7:54 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Feb 19, 2013 at 10:40 AM, Johan Helsingius <julf@julf.com> wrote:
In other words, carriers in the USA don't have a monopoly on their support reps not knowing how their own systems work!
They are probably all sharing the same call center in some low-wage country. Made possible by VoIP, of course... :)
Haha - we're our own worst enemy - the OSS telephony geeks are subverted by the work of their own hands! -MC
Julf
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk@hotmail.com Email: shaheryarkh@googlemail.com
Ask them for a list of devices that they support. Once you have that we can probably help you figure out how to fool them into thinking that they are talking to one of those "supported" devices.
Well, the only "supported" devices are ADSL gateways. For anything else, they just give this set-up advice that looks pretty normal to me: SIP number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXXX SIP proxy: sip.telefoniedienst.nl SIP service location: sip.telefoniedienst.nl SIP signaling port: UDP 5060 SIP register service location: tel.telefoniedienst.nl SIP register signaling port: UDP 5060 SIP domain: tel.telefoniedienst.nl URL type coding: SIP Registration interval: 6000 seconds Session expires: 1800 seconds Reserved RTP port range: UDP 16384-32767 Codecs: First negotiated G.729 secondly negotiated G.711 a-law type DTMF mode: PCM Sending INVITE Anonymous: Not permitted Echo Cancellation G.168: In use Voice Activation Detection: Not in use SIP SUBSCRIBES messages: Not permitted Early Media acknowledge: Not in Use This is what works on the ZyXEL P-2601HN0F1 gateway: SIP Account: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXXX Compression Type: G711ALaw, G.729, G.722 SIP Service Provider Name : telfort SIP Local Port : 5060 SIP Server Address : sip.telefoniedienst.nl SIP Server Port : 5060 REGISTER Server Address : sip.telefoniedienst.nl REGISTER Server Port : 5060 SIP Service Domain : tel.telefoniedienst.nl (different than SIP server) RTP Port Range: 16384-32767 Expiration Duration: 6000 s Register Re-send timer: 512 s Session Expires: 1800 s Min-SE: 1800 s Enable Early Media: Yes And this is what I have in my Freeswitch SIP gateway profile: <gateway name="telfort"> <param name="username" value="020NNNNNNN"/> <param name="auth-username" value="020NNNNNNN"/> <param name="from-user" value="020NNNNNNN"/> <param name="realm" value="tel.telefoniedienst.nl"/> <param name="from-domain" value="tel.telefoniedienst.nl"/> <param name="proxy" value="sip.telefoniedienst.nl"/> <param name="password" value="XXXXXXXX"/> </gateway> It seems like the registration goes through OK, but when I try calling out, I get 013-02-19 10:00:37.236188 [NOTICE] sofia.c:6385 Hangup sofia/external/0651142219 [CS_EXCHANGE_MEDIA] [CALL_REJECTED]
In cases where the provider is so totally foolish like this you have two choices: * Use their foolishness against them and make them think your FreeSWITCH server is really something else
Well, I tried to claim the problem was with the ADSL gateway as well, and they just kept sending the same setup instructions and advice to reboot the gateway. Not helpful.
* Switch providers
I agree, and I am seriously considering it - problem is that their Internet connection is pretty good and keenly priced, and the VoIP add-on to that is priced aggressively. Julf
On 02/19/2013 10:22 AM, Johan Helsingius wrote:
<gateway name="telfort"> <param name="username" value="020NNNNNNN"/> <param name="auth-username" value="020NNNNNNN"/> <param name="from-user" value="020NNNNNNN"/> <param name="realm" value="tel.telefoniedienst.nl"/> <param name="from-domain" value="tel.telefoniedienst.nl"/> <param name="proxy" value="sip.telefoniedienst.nl"/> <param name="password" value="XXXXXXXX"/> </gateway>
It seems like the registration goes through OK, but when I try calling out, I get
013-02-19 10:00:37.236188 [NOTICE] sofia.c:6385 Hangup sofia/external/0651142219 [CS_EXCHANGE_MEDIA] [CALL_REJECTED]
I can't be of much help here other than that I found an Asterisk config that seems to work at http://www.van-sluis.nl/?p=18 [tiscali] type=peer insecure=very context=tiscali-in username=0201234567 fromuser=0201234567 secret=password qualify=yes host=sip.telefoniedienst.nl fromdomain=tel.telefoniedienst.nl dtmf=inband dtmfmode=inband canreinvite=yes nat=yes The difference with the Asterisk config is the use of realm and proxy in your config. Does your box use/offer alaw (and g729) as specified by the sip provider? Regards, Patrick
participants (8)
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Avi Marcus
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Cal Leeming [Simplicity Media Ltd]
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Johan Helsingius
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Michael Collins
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Muhammad Shahzad
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Patrick Lists
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Sergey Okhapkin
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Steven Ayre